update for 2.1.4

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

.gitattributes

@@ -1 +0,0 @@
-*.pnm -diff -text

.gitignore

@@ -15,7 +15,6 @@
 *.pdb
 *.so
 *.so.*
-*.swp
 *.ver
 *-example
 *-test
@@ -28,6 +27,7 @@
 /ffserver
 /config.*
 /coverage.info
+/version.h
 /doc/*.1
 /doc/*.3
 /doc/*.html
@@ -35,36 +35,27 @@
 /doc/config.texi
 /doc/avoptions_codec.texi
 /doc/avoptions_format.texi
-/doc/doxy/html/
-/doc/examples/avio_reading
 /doc/examples/decoding_encoding
-/doc/examples/demuxing_decoding
-/doc/examples/extract_mvs
-/doc/examples/filter_audio
+/doc/examples/demuxing
 /doc/examples/filtering_audio
 /doc/examples/filtering_video
 /doc/examples/metadata
 /doc/examples/muxing
 /doc/examples/pc-uninstalled
-/doc/examples/remuxing
 /doc/examples/resampling_audio
 /doc/examples/scaling_video
-/doc/examples/transcode_aac
-/doc/examples/transcoding
 /doc/fate.txt
+/doc/doxy/html/
 /doc/print_options
 /lcov/
 /libavcodec/*_tablegen
 /libavcodec/*_tables.c
 /libavcodec/*_tables.h
 /libavutil/avconfig.h
-/libavutil/ffversion.h
 /tests/audiogen
 /tests/base64
 /tests/data/
-/tests/pixfmts.mak
 /tests/rotozoom
-/tests/test_copy.ffmeta
 /tests/tiny_psnr
 /tests/tiny_ssim
 /tests/videogen
@@ -83,8 +74,6 @@
 /tools/pktdumper
 /tools/probetest
 /tools/qt-faststart
-/tools/sidxindex
 /tools/trasher
 /tools/seek_print
-/tools/uncoded_frame
 /tools/zmqsend

Changelog

@@ -1,472 +1,7 @@
 Entries are sorted chronologically from oldest to youngest within each release,
 releases are sorted from youngest to oldest.
 
-version 2.6.5:
-- rtmpcrypt: Do the xtea decryption in little endian mode
-- avformat/matroskadec: Check subtitle stream before dereferencing
-- avformat/utils: Do not init parser if probing is unfinished
-- avcodec/jpeg2000dec: Fix potential integer overflow with tile dimensions
-- avcodec/jpeg2000dec: Check SIZ dimensions to be within the supported range
-- avcodec/jpeg2000: Check comp coords to be within the supported size
-- avcodec/jpeg2000: Use av_image_check_size() in ff_jpeg2000_init_component()
-- avcodec/wmaprodec: Check for overread in decode_packet()
-- avcodec/smacker: Check that the data size is a multiple of a sample vector
-- avcodec/takdec: Skip last p2 sample (which is unused)
-- avcodec/dxtory: Fix input size check in dxtory_decode_v1_410()
-- avcodec/dxtory: Fix input size check in dxtory_decode_v1_420()
-- avcodec/error_resilience: avoid accessing previous or next frames tables beyond height
-- avcodec/dpx: Move need_align to act per line
-- avcodec/flashsv: Check size before updating it
-- avcodec/ivi: Check image dimensions
-- avcodec/utils: Better check for channels in av_get_audio_frame_duration()
-- avcodec/jpeg2000dec: Check for duplicate SIZ marker
-- doc/ffmpeg: Clarify that the sdp_file option requires an rtp output.
-- ffmpeg: Don't try and write sdp info if none of the outputs had an rtp format.
-- jvdec: avoid unsigned overflow in comparison
-- avcodec/jpeg2000dec: Clip all tile coordinates
-- avcodec/hevc_ps: Check chroma_format_idc
-- avcodec/microdvddec: Check for string end in 'P' case
-- avcodec/dirac_parser: Fix undefined memcpy() use
-- avformat/xmv: Discard remainder of packet on error
-- avformat/xmv: factor return check out of if/else
-- avcodec/mpeg12dec: Do not call show_bits() with invalid bits
-- libavutil/channel_layout: Check strtol*() for failure
-- avcodec/ffv1dec: Check for 0 quant tables
-- avcodec/mjpegdec: Reinitialize IDCT on BPP changes
-- avcodec/mjpegdec: Check index in ljpeg_decode_yuv_scan() before using it
-- avutil/file_open: avoid file handle inheritance on Windows
-- opusdec: Don't run vector_fmul_scalar on zero length arrays
-- avcodec/ffv1: Initialize vlc_state on allocation
-- avcodec/ffv1dec: update progress in case of broken pointer chains
-- avcodec/ffv1dec: Clear slice coordinates if they are invalid or slice header decoding fails for other reasons
-- avformat/httpauth: Add space after commas in HTTP/RTSP auth header
-- avcodec/x86/sbrdsp: Fix using uninitialized upper 32bit of noise
-- avcodec/ffv1dec: Fix off by 1 error in quant_table_count check
-- avcodec/ffv1dec: Explicitly check read_quant_table() return value
-- avcodec/rangecoder: Check e
-- lavf/webvttenc: Require webvtt file to contain exactly one WebVTT stream.
-- avcodec/mjpegdec: Fix decoding RGBA RCT LJPEG
-- avfilter/af_asyncts: use llabs for int64_t
-- avcodec/g2meet: Also clear tile dimensions on header_fail
-- avcodec/g2meet: Fix potential overflow in tile dimensions check
-- avcodec/svq1dec: Check init_get_bits8() for failure
-- avcodec/tta: Check init_get_bits8() for failure
-- avcodec/vp3: Check init_get_bits8() for failure
-- swresample/swresample: Fix integer overflow in seed calculation
-- avformat/mov: Fix integer overflow in FFABS
-- avutil/common: Add FFNABS()
-- avutil/common: Document FFABS() corner case
-- avformat/dump: Fix integer overflow in aspect ratio calculation
-- avcodec/truemotion1: Check for even width
-- avcodec/mpeg12dec: Set dimensions in mpeg1_decode_sequence() only in absence of errors
-- avcodec/libopusenc: Fix infinite loop on flushing after 0 input
-- avformat/hevc: Check num_long_term_ref_pics_sps to avoid potentially long loops
-- avformat/hevc: Fix parsing errors
-- ffmpeg: Use correct codec_id for av_parser_change() check
-- ffmpeg: Check av_parser_change() for failure
-- ffmpeg: Check for RAWVIDEO and do not relay only on AVFMT_RAWPICTURE
-- ffmpeg: check avpicture_fill() return value
-- avformat/mux: Update sidedata in ff_write_chained()
-- avcodec/flashsvenc: Correct max dimension in error message
-- avcodec/svq1enc: Check dimensions
-- avcodec/dcaenc: clear bitstream end
-- libavcodec/aacdec_template: Use init_get_bits8() in aac_decode_frame()
-- rawdec: fix mjpeg probing buffer size check
-- rawdec: fix mjpeg probing
-- videodsp: don't overread edges in vfix3 emu_edge.
-- lavf/matroskadec: Fully parse and repack MP3 packets
-- avcodec/h264_mp4toannexb_bsf: Reorder operations in nal_size check
-- avformat/oggenc: Check segments_count for headers too
-- avformat/avidec: Workaround broken initial frame
-- hevc: properly handle no_rasl_output_flag when removing pictures from the DPB
-- hevc: fix wpp threading deadlock.
-- avcodec/ffv1: separate slice_count from max_slice_count
-- lavf/img2dec: Fix memory leak
-- avcodec/mp3: fix skipping zeros
-- avformat/srtdec: make sure we probe a number
-- avformat/srtdec: more lenient first line probing
-- doc: mention libavcodec can decode Opus natively
-- MAINTAINERS: Remove myself as leader
-
-
-version 2.6.4:
-- imc: use correct position for flcoeffs2 calculation
-- hevc: check slice address length
-- snow: remove an obsolete av_assert2
-- webp: fix infinite loop in webp_decode_frame
-- wavpack: limit extra_bits to 32 and use get_bits_long
-- ffmpeg: only count got_output/errors in decode_error_stat
-- ffmpeg: exit_on_error if decoding a packet failed
-- pthread_frame: forward error codes when flushing
-- huffyuvdec: validate image size
-- wavpack: use get_bits_long to read up to 32 bits
-- nutdec: check maxpos in read_sm_data before returning success
-- vc1dec: use get_bits_long and limit the read bits to 32
-- mpegaudiodec: copy AVFloatDSPContext from first context to all contexts
-- avcodec/vp8: Check buffer size in vp8_decode_frame_header()
-- avcodec/vp8: Fix null pointer dereference in ff_vp8_decode_free()
-- avcodec/diracdec: Check for hpel_base allocation failure
-- avcodec/rv34: Clear pointers in ff_rv34_decode_init_thread_copy()
-- avfilter/af_aresample: Check ff_all_* for allocation failures
-- avcodec/pthread_frame: clear priv_data, avoid stale pointer in error case
-- swscale/utils: Clear pix buffers
-- avutil/fifo: Fix the case where func() returns less bytes than requested in av_fifo_generic_write()
-- ffmpeg: Fix cleanup after failed allocation of output_files
-- avformat/mov: Fix deallocation when MOVStreamContext failed to allocate
-- ffmpeg: Fix crash with ost->last_frame allocation failure
-- ffmpeg: Fix cleanup with ost = NULL
-- avcodec/pthread_frame: check avctx on deallocation
-- avcodec/sanm: Reset sizes in destroy_buffers()
-- avcodec/alac: Clear pointers in allocate_buffers()
-- bytestream2: set the reader to the end when reading more than available
-- avcodec/utils: use a minimum 32pixel width in  avcodec_align_dimensions2() for H.264
-- avcodec/mpegvideo: Clear pointers in ff_mpv_common_init()
-- oggparsedirac: check return value of init_get_bits
-- wmalosslessdec: reset frame->nb_samples on packet loss
-- wmalosslessdec: avoid reading 0 bits with get_bits
-- avcodec/rawenc: Use ff_alloc_packet() instead of ff_alloc_packet2()
-- avcodec/aacsbr: Assert that bs_num_env is positive
-- avcodec/aacsbr: check that the element type matches before applying SBR
-- avcodec/h264_slice: Use w/h from the AVFrame instead of mb_w/h
-- vp9/update_prob: prevent out of bounds table read
-- avfilter/vf_transpose: Fix rounding error
-- avcodec/pngdec: Check values before updating context in decode_fctl_chunk()
-- avcodec/pngdec: Require a IHDR chunk before fctl
-- avcodec/pngdec: Only allow one IHDR chunk
-- wmavoice: limit wmavoice_decode_packet return value to packet size
-- swscale/swscale_unscaled: Fix rounding difference with RGBA output between little and big endian
-- ffmpeg: Do not use the data/size of a bitstream filter after failure
-- swscale/x86/rgb2rgb_template: fix signedness of v in shuffle_bytes_2103_{mmx,mmxext}
-- swscale/x86/rgb2rgb_template: add missing xmm clobbers
-- vda: unlock the pixel buffer base address.
-- swscale/rgb2rgb_template: Fix signedness of v in shuffle_bytes_2103_c()
-- swscale/rgb2rgb_template: Implement shuffle_bytes_0321_c and fix shuffle_bytes_2103_c on BE
-- swscale/rgb2rgb_template: Disable shuffle_bytes_2103_c on big endian
-- swr: Remember previously set int_sample_format from user
-- matroskadec: check audio sample rate
-- matroskadec: validate audio channels and bitdepth
-- avcodec/dpxenc: implement write16/32 as functions
-- postproc: fix unaligned access
-- ffmpeg: Free last_frame instead of just unref
-- avio: fix potential crashes when combining ffio_ensure_seekback + crc
-- examples/demuxing_decoding: use properties from frame instead of video_dec_ctx
-- h264: er: Copy from the previous reference only if compatible
-- sonic: set avctx->channels in sonic_decode_init
-- vp8: change mv_{min,max}.{x,y} type to int
-- vp9: change type of tile_size from unsigned to int64_t
-- arm: only enable setend on ARMv6
-- libopenjpegdec: check existence of image component data
-- mov: abort on EOF in ff_mov_read_chan
-- ffmpeg_opt: Check for localtime() failure
-- avformat: Fix bug in parse_rps for HEVC.
-- takdec: ensure chan2 is a valid channel index
-- avcodec/h264_slice: Use AVFrame diemensions for grayscale handling
-- avdevice/lavfi: do not rescale AV_NOPTS_VALUE in lavfi_read_packet()
-- libavutil/channel_layout: Correctly return layout when channel specification ends with a trailing 'c'.
-- avcodec/jpeg2000dec: Check that coords match before applying ICT
-- avformat/ffmdec: Check ffio_set_buf_size() return value
-- avcodec/adpcm: Check for overreads
-- avcodec/alsdec: Check for overread
-- avcodec/atrac3plusdec: consume only as many bytes as available
-- libavutil/softfloat: Fix av_normalize1_sf bias.
-- swresample/swresample: Cleanup on init failure.
-- Revert "avformat/rtpenc: check av_packet_get_side_data() return, fix null ptr dereference"
-- avformat/mxfenc: Accept MXF D-10 with 49.999840 Mbit/sec
-- swresample/dither: check memory allocation
-- libopenjpegenc: add NULL check for img before accessing it
-- swresample: Check the return value of resampler->init()
-- h264: Make sure reinit failures mark the context as not initialized
-- avfilter/x86/vf_fspp: Fix invalid combination of opcode and operands
-- ffmpeg_opt: Set the video VBV parameters only for the video stream from -target
-- avcodec/bitstream: Assert that there is enough space left in avpriv_copy_bits()
-- avcodec/put_bits: Assert that there is enough space left in skip_put_bytes()
-- avcodec/mpegvideo_enc: Update the buffer size as more slices are merged
-- avcodec/put_bits: Update size_in_bits in set_put_bits_buffer_size()
-- avformat/wavdec: Increase dts packet threshold to fix more misdetections
-- avformat/wavdec: Increase probe_packets limit
-- nutdec: abort if EOF is reached in decode_info_header/read_sm_data
-- nutdec: stop skipping bytes at EOF
-- nutdec: fix infinite resync loops
-- avformat/nutdec: Check X in 2nd branch of index reading
-- avformat/nutdec: Fix recovery when immedeately after seeking a failure happens
-- avformat/nutdec: Return error on EOF from get_str()
-- rtsp: Make sure we don't write too many transport entries into a fixed-size array
-- rtpenc_jpeg: handle case of picture dimensions not dividing by 8
-- avcodec/golomb: get_ur_golomb_jpegls: Fix reading huge k values
-- avformat/swfdec: Do not error out on pixel format changes
-- avformat/mov: Mark avio context of decompressed atoms as seekable
-- avcodec/mjpegenc_common: Use ff_mpv_reallocate_putbitbuffer()
-- avcodec/mpegvideo: Factor ff_mpv_reallocate_putbitbuffer() out
-- avfilter/x86/vf_hqdn3d: Fix register types
-- avcodec/exr: fix crash caused by merge
-- avcodec/x86/h264_weight: handle weight1=128
-- avcodec/dvbsubdec: Fix buf_size check in dvbsub_parse_display_definition_segment()
-- avcodec/hevc_ps: Only discard overread VPS if a previous is available
-- avcodec/flacenc: Fix Invalid Rice order
-- lavd/xcbgrab: fix comparison with screen size.
-
-version 2.6.3:
-- avcodec/libtheoraenc: Check for av_malloc failure
-- ffmpeg_opt: Fix -timestamp parsing
-- hevc: make avcodec_decode_video2() fail if get_format() fails
-- avcodec/cavsdec: Use ff_set_dimensions()
-- swr: fix alignment issue caused by 8ch sse functions
-- avcodec/mjpegdec: fix len computation in ff_mjpeg_decode_dqt()
-- avcodec/jpeg2000dec: fix boolean operator
-- avcodec/hevc_ps: Explicitly check num_tile_* for negative values
-- avformat/matroskadec: Cleanup error handling for bz2 & zlib
-- avformat/nutdec: Fix use of uinitialized value
-- tools/graph2dot: use larger data types than int for array/string sizes
-- avformat/matroskaenc: Check ff_vorbiscomment_length in put_flac_codecpriv()
-- avcodec/mpeg12dec: use the correct dimensions for checking SAR
-- xcbgrab: Validate the capture area
-- xcbgrab: Do not assume the non shm image data is always available
-- avfilter/lavfutils: disable frame threads when decoding a single image
-- avformat/mov: Do not read ACLR into extradata for H.264
-- ffmpeg: remove incorrect network deinit
-- OpenCL: Avoid potential buffer overflow in cmdutils_opencl.c
-- libvpxenc: only set noise reduction w/vp8
-- vp9: remove another optimization branch in iadst16 which causes overflows.
-- lavf: Reset global flag on deinit
-- network: Do not leave context locked on error
-- vp9: remove one optimization branch in iadst16 which causes overflows.
-- fate: Include branch information in the payload header
-- avformat/utils: Ensure that AVFMT_FLAG_CUSTOM_IO is set before use
-- avformat/img2dec: do not rewind custom io buffers
-- avcodec/alsdec: Use av_mallocz_array() for chan_data to ensure the arrays never contain random data
-- avcodec/atrac3plusdsp: fix on stack alignment
-- swresample/swresample-test: Randomly wipe out channel counts
-- swresample: Check channel layouts and channels against each other and print human readable error messages
-- swresample: Allow reinitialization without ever setting channel layouts (cherry picked from commit 80a28c7509a11114e1aea5b208d56c6646d69c07)
-- swresample: Allow reinitialization without ever setting channel counts
-- dashenc: replace attribute id with contentType for the AdaptationSet element
-- avformat/matroskaenc: Use avoid_negative_ts_use_pts if no stream writes dts
-- avformat/mux: Add avoid_negative_ts_use_pts
-- tests/fate-run: do not attempt to parse tiny_psnrs output if it failed
-- cafdec: free extradata before allocating it
-- imgutils: initialize palette padding bytes in av_image_alloc
-- aacdec: don't return frames without data
-- id3v2: catch avio_read errors in check_tag
-- avi: Validate sample_size
-- aacsbr: break infinite loop in sbr_hf_calc_npatches
-- diracdec: avoid overflow of bytes*8 in decode_lowdelay
-- diracdec: prevent overflow in data_unit_size check
-- avformat/matroskadec: Use tracks[k]->stream instead of s->streams[k]
-- matroskadec: use uint64_t instead of int for index_scale
-- pngdec: don't use AV_PIX_FMT_MONOBLACK for apng
-- pngdec: return correct error code from decode_frame_common
-- nutdec: fix illegal count check in decode_main_header
-- nutdec: fix memleaks on error in nut_read_header
-- apedec: prevent out of array writes in decode_array_0000
-- apedec: set s->samples only when init_frame_decoder succeeded
-- swscale/ppc/swscale_altivec.c: POWER LE support in yuv2planeX_8() delete macro GET_VF() it was wrong
-- alac: reject rice_limit 0 if compression is used
-- alsdec: only adapt order for positive max_order
-- bink: check vst->index_entries before using it
-- mpeg4videodec: only allow a positive length
-- aacpsy: correct calculation of minath in psy_3gpp_init
-- alsdec: validate time diff index
-- alsdec: ensure channel reordering is reversible
-- ac3: validate end in ff_ac3_bit_alloc_calc_mask
-- aacpsy: avoid psy_band->threshold becoming NaN
-- aasc: return correct buffer size from aasc_decode_frame
-- matroskadec: export cover art correctly
-- mxfenc: don't try to write footer without header
-- mxfenc: fix memleaks in mxf_write_footer
-- rtpenc_mpegts: Set chain->rtp_ctx only after avformat_write_header succeeded
-- rtpenc_mpegts: Free the right ->pb in the error path in the init function
-
-version 2.6.2:
-- avcodec/h264: Do not fail with randomly truncated VUIs
-- avcodec/h264_ps: Move truncation check from VUI to SPS
-- avcodec/h264: Be more tolerant to changing pps id between slices
-- avcodec/aacdec: Fix storing state before PCE decode
-- avcodec/h264: reset the counts in the correct context
-- avcodec/h264_slice: Do not reset mb_aff_frame per slice
-- avcodec/h264: finish previous slices before switching to single thread mode
-- avcodec/h264: Fix race between slices where one overwrites data from the next
-- avformat/utils: avoid discarded streams in av_find_default_stream_index()
-- ffmpeg: Fix extradata allocation
-- avcodec/h264_refs: Do not set reference to things which do not exist
-- avcodec/h264: Fail for invalid mixed IDR / non IDR frames in slice threading mode
-- Revert "avcodec/exr: fix memset first arg in reverse_lut()"
-- h264: avoid unnecessary calls to get_format
-- avutil/pca: Check for av_malloc* failures
-- avutil/cpu: add missing check for mmxext to av_force_cpu_flags
-- lavc/dnxhd: Fix pix_fmt change.
-- avformat/http: replace cookies with updated values instead of appending forever
-- avformat/hls: store cookies returned in HLS key response
-- avformat/rmdec: fix support for 0 sized mdpr
-- avcodec/msrledec: restructure msrle_decode_pal4() based on the line number instead of the pixel pointer
-- avcodec/hevc_ps: Check cropping parameters more correctly
-- hevc: make the crop sizes unsigned
-- avcodec/dnxhddec: Reset is_444 if format is not 444
-- avcodec/dnxhddec: Check that the frame is interlaced before using cur_field
-- mips/float_dsp: fix vector_fmul_window_mips on mips64
-- doc: Remove non-existing decklink options.
-
-version 2.6.1:
-- avformat/mov: Disallow ".." in dref unless use_absolute_path is set
-- avfilter/palettegen: make sure at least one frame was sent to the filter
-- avformat/mov: Check for string truncation in mov_open_dref()
-- ac3_fixed: fix out-of-bound read
-- mips/asmdefs: use _ABI64 as defined by gcc
-- hevc: delay ff_thread_finish_setup for hwaccel
-- avcodec/012v: Check dimensions more completely
-- asfenc: fix leaking asf->index_ptr on error
-- roqvideoenc: set enc->avctx in roq_encode_init
-- avcodec/options_table: remove extradata_size from the AVOptions table
-- ffmdec: limit the backward seek to the last resync position
-- Add dependencies to configure file for vf_fftfilt
-- ffmdec: make sure the time base is valid
-- ffmdec: fix infinite loop at EOF
-- ffmdec: initialize f_cprv, f_stvi and f_stau
-- arm: Suppress tags about used cpu arch and extensions
-- mxfdec: Fix the error handling for when strftime fails
-- avcodec/opusdec: Fix delayed sample value
-- avcodec/opusdec: Clear out pointers per packet
-- avcodec/utils: Align YUV411 by as much as the other YUV variants
-- lavc/hevcdsp: Fix compilation for arm with --disable-neon.
-- vp9: fix segmentation map retention with threading enabled.
-- Revert "avutil/opencl: is_compiled flag not being cleared in av_opencl_uninit"
-
-version 2.6:
-- nvenc encoder
-- 10bit spp filter
-- colorlevels filter
-- RIFX format for *.wav files
-- RTP/mpegts muxer
-- non continuous cache protocol support
-- tblend filter
-- cropdetect support for non 8bpp, absolute (if limit >= 1) and relative (if limit < 1.0) threshold
-- Camellia symmetric block cipher
-- OpenH264 encoder wrapper
-- VOC seeking support
-- Closed caption Decoder
-- fspp, uspp, pp7 MPlayer postprocessing filters ported to native filters
-- showpalette filter
-- Twofish symmetric block cipher
-- Support DNx100 (960x720@8)
-- eq2 filter ported from libmpcodecs as eq filter
-- removed libmpcodecs
-- Changed default DNxHD colour range in QuickTime .mov derivatives to mpeg range
-- ported softpulldown filter from libmpcodecs as repeatfields filter
-- dcshift filter
-- RTP depacketizer for loss tolerant payload format for MP3 audio (RFC 5219)
-- RTP depacketizer for AC3 payload format (RFC 4184)
-- palettegen and paletteuse filters
-- VP9 RTP payload format (draft 0) experimental depacketizer
-- RTP depacketizer for DV (RFC 6469)
-- DXVA2-accelerated HEVC decoding
-- AAC ELD 480 decoding
-- Intel QSV-accelerated H.264 decoding
-- DSS SP decoder and DSS demuxer
-- Fix stsd atom corruption in DNxHD QuickTimes
-- Canopus HQX decoder
-- RTP depacketization of T.140 text (RFC 4103)
-- Port MIPS optimizations to 64-bit
-
-
-version 2.5:
-- HEVC/H.265 RTP payload format (draft v6) packetizer
-- SUP/PGS subtitle demuxer
-- ffprobe -show_pixel_formats option
-- CAST128 symmetric block cipher, ECB mode
-- STL subtitle demuxer and decoder
-- libutvideo YUV 4:2:2 10bit support
-- XCB-based screen-grabber
-- UDP-Lite support (RFC 3828)
-- xBR scaling filter
-- AVFoundation screen capturing support
-- ffserver supports codec private options
-- creating DASH compatible fragmented MP4, MPEG-DASH segmenting muxer
-- WebP muxer with animated WebP support
-- zygoaudio decoding support
-- APNG demuxer
-- postproc visualization support
-
-
-version 2.4:
-- Icecast protocol
-- ported lenscorrection filter from frei0r filter
-- large optimizations in dctdnoiz to make it usable
-- ICY metadata are now requested by default with the HTTP protocol
-- support for using metadata in stream specifiers in fftools
-- LZMA compression support in TIFF decoder
-- H.261 RTP payload format (RFC 4587) depacketizer and experimental packetizer
-- HEVC/H.265 RTP payload format (draft v6) depacketizer
-- added codecview filter to visualize information exported by some codecs
-- Matroska 3D support thorugh side data
-- HTML generation using texi2html is deprecated in favor of makeinfo/texi2any
-- silenceremove filter
-
-
-version 2.3:
-- AC3 fixed-point decoding
-- shuffleplanes filter
-- subfile protocol
-- Phantom Cine demuxer
-- replaygain data export
-- VP7 video decoder
-- Alias PIX image encoder and decoder
-- Improvements to the BRender PIX image decoder
-- Improvements to the XBM decoder
-- QTKit input device
-- improvements to OpenEXR image decoder
-- support decoding 16-bit RLE SGI images
-- GDI screen grabbing for Windows
-- alternative rendition support for HTTP Live Streaming
-- AVFoundation input device
-- Direct Stream Digital (DSD) decoder
-- Magic Lantern Video (MLV) demuxer
-- On2 AVC (Audio for Video) decoder
-- support for decoding through DXVA2 in ffmpeg
-- libbs2b-based stereo-to-binaural audio filter
-- libx264 reference frames count limiting depending on level
-- native Opus decoder
-- display matrix export and rotation API
-- WebVTT encoder
-- showcqt multimedia filter
-- zoompan filter
-- signalstats filter
-- hqx filter (hq2x, hq3x, hq4x)
-- flanger filter
-- Image format auto-detection
-- LRC demuxer and muxer
-- Samba protocol (via libsmbclient)
-- WebM DASH Manifest muxer
-- libfribidi support in drawtext
-
-
-version 2.2:
-
-- HNM version 4 demuxer and video decoder
-- Live HDS muxer
-- setsar/setdar filters now support variables in ratio expressions
-- elbg filter
-- string validation in ffprobe
-- support for decoding through VDPAU in ffmpeg (the -hwaccel option)
-- complete Voxware MetaSound decoder
-- remove mp3_header_compress bitstream filter
-- Windows resource files for shared libraries
-- aeval filter
-- stereoscopic 3d metadata handling
-- WebP encoding via libwebp
-- ATRAC3+ decoder
-- VP8 in Ogg demuxing
-- side & metadata support in NUT
-- framepack filter
-- XYZ12 rawvideo support in NUT
-- Exif metadata support in WebP decoder
-- OpenGL device
-- Use metadata_header_padding to control padding in ID3 tags (currently used in
-  MP3, AIFF, and OMA files), FLAC header, and the AVI "junk" block.
-- Mirillis FIC video decoder
-- Support DNx444
-- libx265 encoder
-- dejudder filter
-- Autodetect VDA like all other hardware accelerations
-- aliases and defaults for Ogg subtypes (opus, spx)
+version <next>
 
 
 version 2.1:
@@ -511,8 +46,7 @@ version 2.1:
 - ReplayGain scanner
 - Enhanced Low Delay AAC (ER AAC ELD) decoding (no LD SBR support)
 - Linux framebuffer output device
-- HEVC decoder
-- raw HEVC, HEVC in MOV/MP4, HEVC in Matroska, HEVC in MPEG-TS demuxing
+- HEVC decoder, raw HEVC demuxer, HEVC demuxing in TS, Matroska and MP4
 - mergeplanes filter
 
 
@@ -528,7 +62,7 @@ version 2.0:
 - 10% faster aac encoding on x86 and MIPS
 - sine audio filter source
 - WebP demuxing and decoding support
-- ffmpeg options -filter_script and -filter_complex_script, which allow a
+- new ffmpeg options -filter_script and -filter_complex_script, which allow a
   filtergraph description to be read from a file
 - OpenCL support
 - audio phaser filter
@@ -536,7 +70,7 @@ version 2.0:
 - libquvi demuxer
 - uniform options syntax across all filters
 - telecine filter
-- interlace filter
+- new interlace filter
 - smptehdbars source
 - inverse telecine filters (fieldmatch and decimate)
 - colorbalance filter
@@ -654,7 +188,7 @@ version 1.1:
 - JSON captions for TED talks decoding support
 - SOX Resampler support in libswresample
 - aselect filter
-- SGI RLE 8-bit / Silicon Graphics RLE 8-bit video decoder
+- SGI RLE 8-bit decoder
 - Silicon Graphics Motion Video Compressor 1 & 2 decoder
 - Silicon Graphics Movie demuxer
 - apad filter
@@ -698,9 +232,7 @@ version 1.0:
 - RTMPE protocol support
 - RTMPTE protocol support
 - showwaves and showspectrum filter
-- LucasArts SMUSH SANM playback support
-- LucasArts SMUSH VIMA audio decoder (ADPCM)
-- LucasArts SMUSH demuxer
+- LucasArts SMUSH playback support
 - SAMI, RealText and SubViewer demuxers and decoders
 - Heart Of Darkness PAF playback support
 - iec61883 device
@@ -824,7 +356,6 @@ version 0.10:
 - ffwavesynth decoder
 - aviocat tool
 - ffeval tool
-- support encoding and decoding 4-channel SGI images
 
 
 version 0.9:

INSTALL

@@ -0,0 +1,15 @@
+
+1) Type './configure' to create the configuration. A list of configure
+options is printed by running 'configure --help'.
+
+'configure' can be launched from a directory different from the FFmpeg
+sources to build the objects out of tree. To do this, use an absolute
+path when launching 'configure', e.g. '/ffmpegdir/ffmpeg/configure'.
+
+2) Then type 'make' to build FFmpeg. GNU Make 3.81 or later is required.
+
+3) Type 'make install' to install all binaries and libraries you built.
+
+NOTICE
+
+ - Non system dependencies (e.g. libx264, libvpx) are disabled by default.

INSTALL.md

@@ -1,17 +0,0 @@
-#Installing FFmpeg:
-
-1. Type `./configure` to create the configuration. A list of configure
-options is printed by running `configure --help`.
-
-    `configure` can be launched from a directory different from the FFmpeg
-sources to build the objects out of tree. To do this, use an absolute
-path when launching `configure`, e.g. `/ffmpegdir/ffmpeg/configure`.
-
-2. Then type `make` to build FFmpeg. GNU Make 3.81 or later is required.
-
-3. Type `make install` to install all binaries and libraries you built.
-
-NOTICE
-------
-
- - Non system dependencies (e.g. libx264, libvpx) are disabled by default.

LICENSE

@@ -1,73 +1,68 @@
-#FFmpeg:
+FFmpeg:
 
 Most files in FFmpeg are under the GNU Lesser General Public License version 2.1
-or later (LGPL v2.1+). Read the file `COPYING.LGPLv2.1` for details. Some other
+or later (LGPL v2.1+). Read the file COPYING.LGPLv2.1 for details. Some other
 files have MIT/X11/BSD-style licenses. In combination the LGPL v2.1+ applies to
 FFmpeg.
 
 Some optional parts of FFmpeg are licensed under the GNU General Public License
-version 2 or later (GPL v2+). See the file `COPYING.GPLv2` for details. None of
-these parts are used by default, you have to explicitly pass `--enable-gpl` to
+version 2 or later (GPL v2+). See the file COPYING.GPLv2 for details. None of
+these parts are used by default, you have to explicitly pass --enable-gpl to
 configure to activate them. In this case, FFmpeg's license changes to GPL v2+.
 
-Specifically, the GPL parts of FFmpeg are:
+Specifically, the GPL parts of FFmpeg are
 
 - libpostproc
+- libmpcodecs
 - optional x86 optimizations in the files
-  - `libavcodec/x86/flac_dsp_gpl.asm`
-  - `libavcodec/x86/idct_mmx.c`
+  libavcodec/x86/idct_mmx.c
 - libutvideo encoding/decoding wrappers in
-  `libavcodec/libutvideo*.cpp`
-- the X11 grabber in `libavdevice/x11grab.c`
+  libavcodec/libutvideo*.cpp
+- the X11 grabber in libavdevice/x11grab.c
 - the swresample test app in
-  `libswresample/swresample-test.c`
-- the `texi2pod.pl` tool
+  libswresample/swresample-test.c
+- the texi2pod.pl tool
 - the following filters in libavfilter:
-    - `f_ebur128.c`
-    - `vf_blackframe.c`
-    - `vf_boxblur.c`
-    - `vf_colormatrix.c`
-    - `vf_cropdetect.c`
-    - `vf_delogo.c`
-    - `vf_eq.c`
-    - `vf_fspp.c`
-    - `vf_geq.c`
-    - `vf_histeq.c`
-    - `vf_hqdn3d.c`
-    - `vf_interlace.c`
-    - `vf_kerndeint.c`
-    - `vf_mcdeint.c`
-    - `vf_mpdecimate.c`
-    - `vf_owdenoise.c`
-    - `vf_perspective.c`
-    - `vf_phase.c`
-    - `vf_pp.c`
-    - `vf_pp7.c`
-    - `vf_pullup.c`
-    - `vf_sab.c`
-    - `vf_smartblur.c`
-    - `vf_repeatfields.c`
-    - `vf_spp.c`
-    - `vf_stereo3d.c`
-    - `vf_super2xsai.c`
-    - `vf_tinterlace.c`
-    - `vf_uspp.c`
-    - `vsrc_mptestsrc.c`
+    - f_ebur128.c
+    - vf_blackframe.c
+    - vf_boxblur.c
+    - vf_colormatrix.c
+    - vf_cropdetect.c
+    - vf_decimate.c
+    - vf_delogo.c
+    - vf_geq.c
+    - vf_histeq.c
+    - vf_hqdn3d.c
+    - vf_kerndeint.c
+    - vf_mcdeint.c
+    - vf_mp.c
+    - vf_owdenoise.c
+    - vf_perspective.c
+    - vf_phase.c
+    - vf_pp.c
+    - vf_pullup.c
+    - vf_sab.c
+    - vf_smartblur.c
+    - vf_spp.c
+    - vf_stereo3d.c
+    - vf_super2xsai.c
+    - vf_tinterlace.c
+    - vf_yadif.c
+    - vsrc_mptestsrc.c
 
 Should you, for whatever reason, prefer to use version 3 of the (L)GPL, then
-the configure parameter `--enable-version3` will activate this licensing option
-for you. Read the file `COPYING.LGPLv3` or, if you have enabled GPL parts,
-`COPYING.GPLv3` to learn the exact legal terms that apply in this case.
+the configure parameter --enable-version3 will activate this licensing option
+for you. Read the file COPYING.LGPLv3 or, if you have enabled GPL parts,
+COPYING.GPLv3 to learn the exact legal terms that apply in this case.
 
 There are a handful of files under other licensing terms, namely:
 
-* The files `libavcodec/jfdctfst.c`, `libavcodec/jfdctint_template.c` and
-  `libavcodec/jrevdct.c` are taken from libjpeg, see the top of the files for
+* The files libavcodec/jfdctfst.c, libavcodec/jfdctint_template.c and
+  libavcodec/jrevdct.c are taken from libjpeg, see the top of the files for
   licensing details. Specifically note that you must credit the IJG in the
   documentation accompanying your program if you only distribute executables.
   You must also indicate any changes including additions and deletions to
   those three files in the documentation.
-* `tests/reference.pnm` is under the expat license.
 
 
 external libraries
@@ -85,17 +80,15 @@ The following libraries are under GPL:
     - libutvideo
     - libvidstab
     - libx264
-- libx265
     - libxavs
     - libxvid
-
 When combining them with FFmpeg, FFmpeg needs to be licensed as GPL as well by
-passing `--enable-gpl` to configure.
+passing --enable-gpl to configure.
 
 The OpenCORE and VisualOn libraries are under the Apache License 2.0. That
 license is incompatible with the LGPL v2.1 and the GPL v2, but not with
 version 3 of those licenses. So to combine these libraries with FFmpeg, the
-license version needs to be upgraded by passing `--enable-version3` to configure.
+license version needs to be upgraded by passing --enable-version3 to configure.
 
 incompatible libraries
 ----------------------
@@ -103,7 +96,7 @@ incompatible libraries
 The Fraunhofer AAC library, FAAC and aacplus are under licenses which
 are incompatible with the GPLv2 and v3. We do not know for certain if their
 licenses are compatible with the LGPL.
-If you wish to enable these libraries, pass `--enable-nonfree` to configure.
+If you wish to enable these libraries, pass --enable-nonfree to configure.
 But note that if you enable any of these libraries the resulting binary will
 be under a complex license mix that is more restrictive than the LGPL and that
 may result in additional obligations. It is possible that these

MAINTAINERS

@@ -14,6 +14,7 @@ patches and related discussions.
 Project Leader
 ==============
 
+Michael Niedermayer
   final design decisions
 
 
@@ -30,7 +31,7 @@ ffprobe:
   ffprobe.c                             Stefano Sabatini
 
 ffserver:
-  ffserver.c                            Reynaldo H. Verdejo Pinochet
+  ffserver.c, ffserver.h                Baptiste Coudurier
 
 Commandline utility code:
   cmdutils.c, cmdutils.h                Michael Niedermayer
@@ -43,8 +44,8 @@ Miscellaneous Areas
 ===================
 
 documentation                           Stefano Sabatini, Mike Melanson, Timothy Gu
-build system (configure, makefiles)     Diego Biurrun, Mans Rullgard
-project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser
+build system (configure,Makefiles)      Diego Biurrun, Mans Rullgard
+project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Döffinger, Alexander Strasser
 presets                                 Robert Swain
 metadata subsystem                      Aurelien Jacobs
 release management                      Michael Niedermayer
@@ -53,9 +54,7 @@ release management                      Michael Niedermayer
 Communication
 =============
 
-website                                 Deby Barbara Lepage
-fate.ffmpeg.org                         Timothy Gu
-Trac bug tracker                        Alexander Strasser, Michael Niedermayer, Carl Eugen Hoyos, Lou Logan
+website                                 Robert Swain, Lou Logan
 mailinglists                            Michael Niedermayer, Baptiste Coudurier, Lou Logan
 Google+                                 Paul B Mahol, Michael Niedermayer, Alexander Strasser
 Twitter                                 Lou Logan
@@ -74,7 +73,6 @@ Other:
   bprint                                Nicolas George
   bswap.h
   des                                   Reimar Doeffinger
-  dynarray.h                            Nicolas George
   eval.c, eval.h                        Michael Niedermayer
   float_dsp                             Loren Merritt
   hash                                  Reimar Doeffinger
@@ -131,7 +129,6 @@ Generic Parts:
     tableprint.c, tableprint.h          Reimar Doeffinger
   fixed point FFT:
     fft*                                Zeljko Lukac
-  Text Subtitles                        Clément Bœsch
 
 Codecs:
   4xm.c                                 Michael Niedermayer
@@ -145,7 +142,6 @@ Codecs:
   ass*                                  Aurelien Jacobs
   asv*                                  Michael Niedermayer
   atrac3*                               Benjamin Larsson
-  atrac3plus*                           Maxim Poliakovski
   bgmc.c, bgmc.h                        Thilo Borgmann
   bink.c                                Kostya Shishkov
   binkaudio.c                           Peter Ross
@@ -154,8 +150,6 @@ Codecs:
   cdxl.c                                Paul B Mahol
   celp_filters.*                        Vitor Sessak
   cinepak.c                             Roberto Togni
-  cinepakenc.c                          Rl / Aetey G.T. AB
-  ccaption_dec.c                        Anshul Maheshwari
   cljr                                  Alex Beregszaszi
   cllc.c                                Derek Buitenhuis
   cook.c, cookdata.h                    Benjamin Larsson
@@ -165,15 +159,12 @@ Codecs:
   dca.c                                 Kostya Shishkov, Benjamin Larsson
   dnxhd*                                Baptiste Coudurier
   dpcm.c                                Mike Melanson
-  dss_sp.c                              Oleksij Rempel, Michael Niedermayer
   dv.c                                  Roman Shaposhnik
-  dvbsubdec.c                           Anshul Maheshwari
   dxa.c                                 Kostya Shishkov
   eacmv*, eaidct*, eat*                 Peter Ross
   exif.c, exif.h                        Thilo Borgmann
-  ffv1*                                 Michael Niedermayer
+  ffv1.c                                Michael Niedermayer
   ffwavesynth.c                         Nicolas George
-  fic.c                                 Derek Buitenhuis
   flac*                                 Justin Ruggles
   flashsv*                              Benjamin Larsson
   flicvideo.c                           Mike Melanson
@@ -183,7 +174,7 @@ Codecs:
   h261*                                 Michael Niedermayer
   h263*                                 Michael Niedermayer
   h264*                                 Loren Merritt, Michael Niedermayer
-  huffyuv*                              Michael Niedermayer, Christophe Gisquet
+  huffyuv.c                             Michael Niedermayer
   idcinvideo.c                          Mike Melanson
   imc*                                  Benjamin Larsson
   indeo2*                               Kostya Shishkov
@@ -206,11 +197,8 @@ Codecs:
   libtheoraenc.c                        David Conrad
   libutvideo*                           Derek Buitenhuis
   libvorbis.c                           David Conrad
-  libvpx*                               James Zern
   libx264.c                             Mans Rullgard, Jason Garrett-Glaser
-  libx265.c                             Derek Buitenhuis
   libxavs.c                             Stefan Gehrer
-  libzvbi-teletextdec.c                 Marton Balint
   loco.c                                Kostya Shishkov
   lzo.h, lzo.c                          Reimar Doeffinger
   mdec.c                                Michael Niedermayer
@@ -227,7 +215,6 @@ Codecs:
   msvideo1.c                            Mike Melanson
   nellymoserdec.c                       Benjamin Larsson
   nuv.c                                 Reimar Doeffinger
-  nvenc.c                               Timo Rothenpieler
   paf.*                                 Paul B Mahol
   pcx.c                                 Ivo van Poorten
   pgssubdec.c                           Reimar Doeffinger
@@ -244,12 +231,12 @@ Codecs:
   rtjpeg.c, rtjpeg.h                    Reimar Doeffinger
   rv10.c                                Michael Niedermayer
   rv3*                                  Kostya Shishkov
-  rv4*                                  Kostya Shishkov, Christophe Gisquet
+  rv4*                                  Kostya Shishkov
   s3tc*                                 Ivo van Poorten
   smacker.c                             Kostya Shishkov
   smc.c                                 Mike Melanson
   smvjpegdec.c                          Ash Hughes
-  snow*                                 Michael Niedermayer, Loren Merritt
+  snow.c                                Michael Niedermayer, Loren Merritt
   sonic.c                               Alex Beregszaszi
   srt*                                  Aurelien Jacobs
   sunrast.c                             Ivo van Poorten
@@ -268,18 +255,17 @@ Codecs:
   v410*.c                               Derek Buitenhuis
   vb.c                                  Kostya Shishkov
   vble.c                                Derek Buitenhuis
-  vc1*                                  Kostya Shishkov, Christophe Gisquet
+  vc1*                                  Kostya Shishkov
   vcr1.c                                Michael Niedermayer
   vda_h264_dec.c                        Xidorn Quan
   vima.c                                Paul B Mahol
   vmnc.c                                Kostya Shishkov
-  vorbisdec.c                           Denes Balatoni, David Conrad
-  vorbisenc.c                           Oded Shimon
+  vorbis_dec.c                          Denes Balatoni, David Conrad
+  vorbis_enc.c                          Oded Shimon
   vp3*                                  Mike Melanson
   vp5                                   Aurelien Jacobs
   vp6                                   Aurelien Jacobs
   vp8                                   David Conrad, Jason Garrett-Glaser, Ronald Bultje
-  vp9                                   Ronald Bultje, Clément Bœsch
   vqavideo.c                            Mike Melanson
   wavpack.c                             Kostya Shishkov
   wmaprodec.c                           Sascha Sommer
@@ -310,21 +296,15 @@ libavdevice
     libavdevice/avdevice.h
 
 
-  avfoundation.m                        Thilo Borgmann
-  decklink*                             Deti Fliegl
-  dshow.c                               Roger Pack (CC rogerdpack@gmail.com)
+  dshow.c                               Roger Pack
   fbdev_enc.c                           Lukasz Marek
-  gdigrab.c                             Roger Pack (CC rogerdpack@gmail.com)
   iec61883.c                            Georg Lippitsch
   lavfi                                 Stefano Sabatini
   libdc1394.c                           Roman Shaposhnik
-  opengl_enc.c                          Lukasz Marek
   pulse_audio_enc.c                     Lukasz Marek
-  qtkit.m                               Thilo Borgmann
   sdl                                   Stefano Sabatini
-  v4l2.c                                Giorgio Vazzana
+  v4l2.c                                Luca Abeni
   vfwcap.c                              Ramiro Polla
-  xv.c                                  Lukasz Marek
 
 libavfilter
 ===========
@@ -346,20 +326,14 @@ Filters:
   af_compand.c                          Paul B Mahol
   af_ladspa.c                           Paul B Mahol
   af_pan.c                              Nicolas George
-  af_silenceremove.c                    Paul B Mahol
   avf_avectorscope.c                    Paul B Mahol
-  avf_showcqt.c                         Muhammad Faiz
   vf_blend.c                            Paul B Mahol
   vf_colorbalance.c                     Paul B Mahol
-  vf_dejudder.c                         Nicholas Robbins
   vf_delogo.c                           Jean Delvare (CC <khali@linux-fr.org>)
   vf_drawbox.c/drawgrid                 Andrey Utkin
   vf_extractplanes.c                    Paul B Mahol
   vf_histogram.c                        Paul B Mahol
-  vf_hqx.c                              Clément Bœsch
-  vf_idet.c                             Pascal Massimino
   vf_il.c                               Paul B Mahol
-  vf_lenscorrection.c                   Daniel Oberhoff
   vf_mergeplanes.c                      Paul B Mahol
   vf_psnr.c                             Paul B Mahol
   vf_scale.c                            Michael Niedermayer
@@ -388,7 +362,6 @@ Muxers/Demuxers:
   aiffdec.c                             Baptiste Coudurier, Matthieu Bouron
   aiffenc.c                             Baptiste Coudurier, Matthieu Bouron
   ape.c                                 Kostya Shishkov
-  apngdec.c                             Benoit Fouet
   ass*                                  Aurelien Jacobs
   astdec.c                              Paul B Mahol
   astenc.c                              James Almer
@@ -401,7 +374,6 @@ Muxers/Demuxers:
   cdxl.c                                Paul B Mahol
   crc.c                                 Michael Niedermayer
   daud.c                                Reimar Doeffinger
-  dss.c                                 Oleksij Rempel, Michael Niedermayer
   dtshddec.c                            Paul B Mahol
   dv.c                                  Roman Shaposhnik
   dxa.c                                 Kostya Shishkov
@@ -413,7 +385,6 @@ Muxers/Demuxers:
   flvdec.c, flvenc.c                    Michael Niedermayer
   gxf.c                                 Reimar Doeffinger
   gxfenc.c                              Baptiste Coudurier
-  hls.c                                 Anssi Hannula
   idcin.c                               Mike Melanson
   idroqdec.c                            Mike Melanson
   iff.c                                 Jaikrishnan Menon
@@ -431,7 +402,6 @@ Muxers/Demuxers:
   matroska.c                            Aurelien Jacobs
   matroskadec.c                         Aurelien Jacobs
   matroskaenc.c                         David Conrad
-  matroska subtitles (matroskaenc.c)    John Peebles
   metadata*                             Aurelien Jacobs
   mgsts.c                               Paul B Mahol
   microdvd*                             Aurelien Jacobs
@@ -441,15 +411,14 @@ Muxers/Demuxers:
   mpc.c                                 Kostya Shishkov
   mpeg.c                                Michael Niedermayer
   mpegenc.c                             Michael Niedermayer
-  mpegts.c                              Marton Balint
-  mpegtsenc.c                           Baptiste Coudurier
+  mpegts*                               Baptiste Coudurier
   msnwc_tcp.c                           Ramiro Polla
   mtv.c                                 Reynaldo H. Verdejo Pinochet
   mxf*                                  Baptiste Coudurier
   mxfdec.c                              Tomas Härdin
   nistspheredec.c                       Paul B Mahol
   nsvdec.c                              Francois Revol
-  nut*                                  Michael Niedermayer
+  nut.c                                 Michael Niedermayer
   nuv.c                                 Reimar Doeffinger
   oggdec.c, oggdec.h                    David Conrad
   oggenc.c                              Baptiste Coudurier
@@ -466,23 +435,15 @@ Muxers/Demuxers:
   rmdec.c, rmenc.c                      Ronald S. Bultje, Kostya Shishkov
   rtmp*                                 Kostya Shishkov
   rtp.c, rtpenc.c                       Martin Storsjo
-  rtpdec_ac3.*                          Gilles Chanteperdrix
-  rtpdec_dv.*                           Thomas Volkert
-  rtpdec_h261.*, rtpenc_h261.*          Thomas Volkert
-  rtpdec_hevc.*, rtpenc_hevc.*          Thomas Volkert
-  rtpdec_mpa_robust.*                   Gilles Chanteperdrix
   rtpdec_asf.*                          Ronald S. Bultje
-  rtpdec_vp9.c                          Thomas Volkert
   rtpenc_mpv.*, rtpenc_aac.*            Martin Storsjo
   rtsp.c                                Luca Barbato
   sbgdec.c                              Nicolas George
   sdp.c                                 Martin Storsjo
   segafilm.c                            Mike Melanson
-  segment.c                             Stefano Sabatini
   siff.c                                Kostya Shishkov
   smacker.c                             Kostya Shishkov
   smjpeg*                               Paul B Mahol
-  spdif*                                Anssi Hannula
   srtdec.c                              Aurelien Jacobs
   swf.c                                 Baptiste Coudurier
   takdec.c                              Paul B Mahol
@@ -491,7 +452,6 @@ Muxers/Demuxers:
   voc.c                                 Aurelien Jacobs
   wav.c                                 Michael Niedermayer
   wc3movie.c                            Mike Melanson
-  webm dash (matroskaenc.c)             Vignesh Venkatasubramanian
   webvtt*                               Matthew J Heaney
   westwood.c                            Mike Melanson
   wtv.c                                 Peter Ross
@@ -505,7 +465,6 @@ Protocols:
   libssh.c                              Lukasz Marek
   mms*.c                                Ronald S. Bultje
   udp.c                                 Luca Abeni
-  icecast.c                             Marvin Scholz
 
 
 libswresample
@@ -534,8 +493,6 @@ Amiga / PowerPC                         Colin Ward
 Linux / PowerPC                         Luca Barbato
 Windows MinGW                           Alex Beregszaszi, Ramiro Polla
 Windows Cygwin                          Victor Paesa
-Windows MSVC                            Matthew Oliver
-Windows ICL                             Matthew Oliver
 ADI/Blackfin DSP                        Marc Hoffman
 Sparc                                   Roman Shaposhnik
 x86                                     Michael Niedermayer
@@ -544,10 +501,8 @@ x86                                     Michael Niedermayer
 Releases
 ========
 
-2.6                                     Michael Niedermayer
-2.5                                     Michael Niedermayer
-2.4                                     Michael Niedermayer
-2.2                                     Michael Niedermayer
+2.1                                     Michael Niedermayer
+2.0                                     Michael Niedermayer
 
 If you want to maintain an older release, please contact us
 
@@ -563,7 +518,7 @@ Attila Kinali                 11F0 F9A6 A1D2 11F6 C745 D10C 6520 BCDD F2DF E765
 Baptiste Coudurier            8D77 134D 20CC 9220 201F C5DB 0AC9 325C 5C1A BAAA
 Ben Littler                   3EE3 3723 E560 3214 A8CD 4DEB 2CDB FCE7 768C 8D2C
 Benoit Fouet                  B22A 4F4F 43EF 636B BB66 FCDC 0023 AE1E 2985 49C8
-Clément Bœsch                 52D0 3A82 D445 F194 DB8B 2B16 87EE 2CB8 F4B8 FCF9
+Bœsch Clément                 52D0 3A82 D445 F194 DB8B 2B16 87EE 2CB8 F4B8 FCF9
 Daniel Verkamp                78A6 07ED 782C 653E C628 B8B9 F0EB 8DD8 2F0E 21C7
 Diego Biurrun                 8227 1E31 B6D9 4994 7427 E220 9CAE D6CC 4757 FCC5
 FFmpeg release signing key    FCF9 86EA 15E6 E293 A564 4F10 B432 2F04 D676 58D8
@@ -578,14 +533,12 @@ Michael Niedermayer           9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
 Nicolas George                24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
 Panagiotis Issaris            6571 13A3 33D9 3726 F728 AA98 F643 B12E ECF3 E029
 Peter Ross                    A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B
-Reimar Doeffinger             C61D 16E5 9E2C D10C 8958 38A4 0899 A2B9 06D4 D9C7
+Reimar Döffinger              C61D 16E5 9E2C D10C 8958 38A4 0899 A2B9 06D4 D9C7
 Reinhard Tartler              9300 5DC2 7E87 6C37 ED7B CA9A 9808 3544 9453 48A4
 Reynaldo H. Verdejo Pinochet  6E27 CD34 170C C78E 4D4F 5F40 C18E 077F 3114 452A
 Robert Swain                  EE7A 56EA 4A81 A7B5 2001 A521 67FA 362D A2FC 3E71
 Sascha Sommer                 38A0 F88B 868E 9D3A 97D4 D6A0 E823 706F 1E07 0D3C
 Stefano Sabatini              0D0B AD6B 5330 BBAD D3D6 6A0C 719C 2839 FC43 2D5F
 Stephan Hilb                  4F38 0B3A 5F39 B99B F505 E562 8D5C 5554 4E17 8863
-Tiancheng "Timothy" Gu        9456 AFC0 814A 8139 E994 8351 7FE6 B095 B582 B0D4
-Tim Nicholson                 38CF DB09 3ED0 F607 8B67 6CED 0C0B FC44 8B0B FC83
 Tomas Härdin                  A79D 4E3D F38F 763F 91F5 8B33 A01E 8AE0 41BB 2551
 Wei Gao                       4269 7741 857A 0E60 9EC5 08D2 4744 4EFA 62C1 87B9

Makefile

@@ -4,47 +4,36 @@ include config.mak
 vpath %.c    $(SRC_PATH)
 vpath %.cpp  $(SRC_PATH)
 vpath %.h    $(SRC_PATH)
-vpath %.m    $(SRC_PATH)
 vpath %.S    $(SRC_PATH)
 vpath %.asm  $(SRC_PATH)
-vpath %.rc   $(SRC_PATH)
 vpath %.v    $(SRC_PATH)
 vpath %.texi $(SRC_PATH)
 vpath %/fate_config.sh.template $(SRC_PATH)
 
-AVPROGS-$(CONFIG_FFMPEG)   += ffmpeg
-AVPROGS-$(CONFIG_FFPLAY)   += ffplay
-AVPROGS-$(CONFIG_FFPROBE)  += ffprobe
-AVPROGS-$(CONFIG_FFSERVER) += ffserver
+PROGS-$(CONFIG_FFMPEG)   += ffmpeg
+PROGS-$(CONFIG_FFPLAY)   += ffplay
+PROGS-$(CONFIG_FFPROBE)  += ffprobe
+PROGS-$(CONFIG_FFSERVER) += ffserver
 
-AVPROGS    := $(AVPROGS-yes:%=%$(PROGSSUF)$(EXESUF))
-INSTPROGS   = $(AVPROGS-yes:%=%$(PROGSSUF)$(EXESUF))
-PROGS      += $(AVPROGS)
-
-AVBASENAMES  = ffmpeg ffplay ffprobe ffserver
-ALLAVPROGS   = $(AVBASENAMES:%=%$(PROGSSUF)$(EXESUF))
-ALLAVPROGS_G = $(AVBASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
-
-$(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog) += cmdutils.o))
-$(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog)-$(CONFIG_OPENCL) += cmdutils_opencl.o))
-
-OBJS-ffmpeg                   += ffmpeg_opt.o ffmpeg_filter.o
-OBJS-ffmpeg-$(HAVE_VDPAU_X11) += ffmpeg_vdpau.o
-OBJS-ffmpeg-$(HAVE_DXVA2_LIB) += ffmpeg_dxva2.o
-OBJS-ffmpeg-$(CONFIG_VDA)     += ffmpeg_vda.o
-OBJS-ffserver                 += ffserver_config.o
+PROGS      := $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))
+INSTPROGS   = $(PROGS-yes:%=%$(PROGSSUF)$(EXESUF))
 
+OBJS        = cmdutils.o $(EXEOBJS)
+OBJS-ffmpeg = ffmpeg_opt.o ffmpeg_filter.o
 TESTTOOLS   = audiogen videogen rotozoom tiny_psnr tiny_ssim base64
 HOSTPROGS  := $(TESTTOOLS:%=tests/%) doc/print_options
-TOOLS       = qt-faststart trasher uncoded_frame
+TOOLS       = qt-faststart trasher
 TOOLS-$(CONFIG_ZLIB) += cws2fws
 
-# $(FFLIBS-yes) needs to be in linking order
+BASENAMES   = ffmpeg ffplay ffprobe ffserver
+ALLPROGS    = $(BASENAMES:%=%$(PROGSSUF)$(EXESUF))
+ALLPROGS_G  = $(BASENAMES:%=%$(PROGSSUF)_g$(EXESUF))
+
 FFLIBS-$(CONFIG_AVDEVICE) += avdevice
 FFLIBS-$(CONFIG_AVFILTER) += avfilter
 FFLIBS-$(CONFIG_AVFORMAT) += avformat
-FFLIBS-$(CONFIG_AVCODEC)    += avcodec
 FFLIBS-$(CONFIG_AVRESAMPLE) += avresample
+FFLIBS-$(CONFIG_AVCODEC)  += avcodec
 FFLIBS-$(CONFIG_POSTPROC) += postproc
 FFLIBS-$(CONFIG_SWRESAMPLE)+= swresample
 FFLIBS-$(CONFIG_SWSCALE)  += swscale
@@ -61,14 +50,16 @@ include $(SRC_PATH)/common.mak
 FF_EXTRALIBS := $(FFEXTRALIBS)
 FF_DEP_LIBS  := $(DEP_LIBS)
 
-all: $(AVPROGS)
+all: $(PROGS)
+
+$(PROGS): %$(EXESUF): %_g$(EXESUF)
+	$(CP) $< $@
+	$(STRIP) $@
 
 $(TOOLS): %$(EXESUF): %.o $(EXEOBJS)
-	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS)
+	$(LD) $(LDFLAGS) $(LD_O) $^ $(ELIBS)
 
 tools/cws2fws$(EXESUF): ELIBS = $(ZLIB)
-tools/uncoded_frame$(EXESUF): $(FF_DEP_LIBS)
-tools/uncoded_frame$(EXESUF): ELIBS = $(FF_EXTRALIBS)
 
 config.h: .config
 .config: $(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c))
@@ -78,10 +69,11 @@ config.h: .config
 
 SUBDIR_VARS := CLEANFILES EXAMPLES FFLIBS HOSTPROGS TESTPROGS TOOLS      \
                HEADERS ARCH_HEADERS BUILT_HEADERS SKIPHEADERS            \
-               ARMV5TE-OBJS ARMV6-OBJS ARMV8-OBJS VFP-OBJS NEON-OBJS     \
-               ALTIVEC-OBJS MMX-OBJS YASM-OBJS                           \
-               MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSPR1-OBJS                \
-               OBJS SLIBOBJS HOSTOBJS TESTOBJS
+               ARMV5TE-OBJS ARMV6-OBJS VFP-OBJS NEON-OBJS                \
+               ALTIVEC-OBJS VIS-OBJS                                     \
+               MMX-OBJS YASM-OBJS                                        \
+               MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSPR1-OBJS MIPS32R2-OBJS  \
+               OBJS HOSTOBJS TESTOBJS
 
 define RESET
 $(1) :=
@@ -93,16 +85,13 @@ $(foreach V,$(SUBDIR_VARS),$(eval $(call RESET,$(V))))
 SUBDIR := $(1)/
 include $(SRC_PATH)/$(1)/Makefile
 -include $(SRC_PATH)/$(1)/$(ARCH)/Makefile
--include $(SRC_PATH)/$(1)/$(INTRINSICS)/Makefile
 include $(SRC_PATH)/library.mak
 endef
 
 $(foreach D,$(FFLIBS),$(eval $(call DOSUBDIR,lib$(D))))
 
-include $(SRC_PATH)/doc/Makefile
-
 define DOPROG
-OBJS-$(1) += $(1).o $(EXEOBJS) $(OBJS-$(1)-yes)
+OBJS-$(1) += $(1).o cmdutils.o $(EXEOBJS)
 $(1)$(PROGSSUF)_g$(EXESUF): $$(OBJS-$(1))
 $$(OBJS-$(1)): CFLAGS  += $(CFLAGS-$(1))
 $(1)$(PROGSSUF)_g$(EXESUF): LDFLAGS += $(LDFLAGS-$(1))
@@ -110,16 +99,10 @@ $(1)$(PROGSSUF)_g$(EXESUF): FF_EXTRALIBS += $(LIBS-$(1))
 -include $$(OBJS-$(1):.o=.d)
 endef
 
-$(foreach P,$(PROGS),$(eval $(call DOPROG,$(P:$(PROGSSUF)$(EXESUF)=))))
-
-ffprobe.o cmdutils.o libavcodec/utils.o libavformat/utils.o libavdevice/avdevice.o libavfilter/avfilter.o libavutil/utils.o libpostproc/postprocess.o libswresample/swresample.o libswscale/utils.o : libavutil/ffversion.h
-
-$(PROGS): %$(PROGSSUF)$(EXESUF): %$(PROGSSUF)_g$(EXESUF)
-	$(CP) $< $@
-	$(STRIP) $@
+$(foreach P,$(PROGS-yes),$(eval $(call DOPROG,$(P))))
 
 %$(PROGSSUF)_g$(EXESUF): %.o $(FF_DEP_LIBS)
-	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
+	$(LD) $(LDFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
 
 OBJDIRS += tools
 
@@ -131,14 +114,14 @@ GIT_LOG     = $(SRC_PATH)/.git/logs/HEAD
 .version: $(wildcard $(GIT_LOG)) $(VERSION_SH) config.mak
 .version: M=@
 
-libavutil/ffversion.h .version:
-	$(M)$(VERSION_SH) $(SRC_PATH) libavutil/ffversion.h $(EXTRA_VERSION)
+version.h .version:
+	$(M)$(VERSION_SH) $(SRC_PATH) version.h $(EXTRA_VERSION)
 	$(Q)touch .version
 
 # force version.sh to run whenever version might have changed
 -include .version
 
-ifdef AVPROGS
+ifdef PROGS
 install: install-progs install-data
 endif
 
@@ -149,7 +132,7 @@ install-libs: install-libs-yes
 install-progs-yes:
 install-progs-$(CONFIG_SHARED): install-libs
 
-install-progs: install-progs-yes $(AVPROGS)
+install-progs: install-progs-yes $(PROGS)
 	$(Q)mkdir -p "$(BINDIR)"
 	$(INSTALL) -c -m 755 $(INSTPROGS) "$(BINDIR)"
 
@@ -161,13 +144,13 @@ install-data: $(DATA_FILES) $(EXAMPLES_FILES)
 uninstall: uninstall-libs uninstall-headers uninstall-progs uninstall-data
 
 uninstall-progs:
-	$(RM) $(addprefix "$(BINDIR)/", $(ALLAVPROGS))
+	$(RM) $(addprefix "$(BINDIR)/", $(ALLPROGS))
 
 uninstall-data:
 	$(RM) -r "$(DATADIR)"
 
 clean::
-	$(RM) $(ALLAVPROGS) $(ALLAVPROGS_G)
+	$(RM) $(ALLPROGS) $(ALLPROGS_G)
 	$(RM) $(CLEANSUFFIXES)
 	$(RM) $(CLEANSUFFIXES:%=tools/%)
 	$(RM) -r coverage-html
@@ -175,13 +158,14 @@ clean::
 
 distclean::
 	$(RM) $(DISTCLEANSUFFIXES)
-	$(RM) config.* .config libavutil/avconfig.h .version version.h libavutil/ffversion.h libavcodec/codec_names.h
+	$(RM) config.* .config libavutil/avconfig.h .version version.h libavcodec/codec_names.h
 
 config:
 	$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)
 
 check: all alltools examples testprogs fate
 
+include $(SRC_PATH)/doc/Makefile
 include $(SRC_PATH)/tests/Makefile
 
 $(sort $(OBJDIRS)):

README

@@ -0,0 +1,18 @@
+FFmpeg README
+-------------
+
+1) Documentation
+----------------
+
+* Read the documentation in the doc/ directory in git.
+  You can also view it online at http://ffmpeg.org/documentation.html
+
+2) Licensing
+------------
+
+* See the LICENSE file.
+
+3) Build and Install
+--------------------
+
+* See the INSTALL file.

README.md

@@ -1,42 +0,0 @@
-FFmpeg README
-=============
-
-FFmpeg is a collection of libraries and tools to process multimedia content
-such as audio, video, subtitles and related metadata.
-
-## Libraries
-
-* `libavcodec` provides implementation of a wider range of codecs.
-* `libavformat` implements streaming protocols, container formats and basic I/O access.
-* `libavutil` includes hashers, decompressors and miscellaneous utility functions.
-* `libavfilter` provides a mean to alter decoded Audio and Video through chain of filters.
-* `libavdevice` provides an abstraction to access capture and playback devices.
-* `libswresample` implements audio mixing and resampling routines.
-* `libswscale` implements color conversion and scaling routines.
-
-## Tools
-
-* [ffmpeg](http://ffmpeg.org/ffmpeg.html) is a command line toolbox to
-  manipulate, convert and stream multimedia content.
-* [ffplay](http://ffmpeg.org/ffplay.html) is a minimalistic multimedia player.
-* [ffprobe](http://ffmpeg.org/ffprobe.html) is a simple analysis tool to inspect
-  multimedia content.
-* [ffserver](http://ffmpeg.org/ffserver.html) is a multimedia streaming server
-  for live broadcasts.
-* Additional small tools such as `aviocat`, `ismindex` and `qt-faststart`.
-
-## Documentation
-
-The offline documentation is available in the **doc/** directory.
-
-The online documentation is available in the main [website](http://ffmpeg.org)
-and in the [wiki](http://trac.ffmpeg.org).
-
-### Examples
-
-Coding examples are available in the **doc/examples** directory.
-
-## License
-
-FFmpeg codebase is mainly LGPL-licensed with optional components licensed under
-GPL. Please refer to the LICENSE file for detailed information.

RELEASE

@@ -1 +1 @@
-2.6.5
+2.1.4

RELEASE_NOTES

@@ -1,65 +0,0 @@
-
-              ┌─────────────────────────────────────────────┐
-              │ RELEASE NOTES for FFmpeg 2.6 "Grothendieck" │
-              └─────────────────────────────────────────────┘
-
-   The FFmpeg Project proudly presents FFmpeg 2.6 "Grothendieck", about 3
-   months after the release of FFmpeg 2.5.
-
-   A lot of important work got in this time, so let's start talking about what
-   we like to brag the most about: features.
-
-   A lot of people will probably be happy to hear that we now have support for
-   NVENC — the Nvidia Video Encoder interface for H.264 encoding — thanks to
-   Timo Rothenpieler, with some little help from NVIDIA and Philip Langdale.
-
-   People in the broadcasting industry might also be interested in the first
-   steps of closed captions support with the introduction of a decoder by
-   Anshul Maheswhwari.
-
-   Regarding filters love, we improved and added many. We could talk about the
-   10-bit support in spp, but maybe it's more important to mention the addition
-   of colorlevels (yet another color handling filter), tblend (allowing you
-   to for example run a diff between successive frames of a video stream), or
-   the dcshift audio filter.
-
-   There are also two other important filters landing in libavfilter: palettegen
-   and paletteuse. Both submitted by the Stupeflix company. These filters will
-   be very useful in case you are looking for creating high quality GIFs, a
-   format that still bravely fights annihilation in 2015.
-
-   There are many other new features, but let's follow-up on one big cleanup
-   achievement: the libmpcodecs (MPlayer filters) wrapper is finally dead. The
-   last remaining filters (softpulldown/repeatfields, eq*, and various
-   postprocessing filters) were ported by Arwa Arif (OPW student) and Paul B
-   Mahol.
-
-   Concerning API changes, there are not many things to mention. Though, the
-   introduction of device inputs and outputs listing by Lukasz Marek is a
-   notable addition (try ffmpeg -sources or ffmpeg -sinks for an example of
-   the usage). As usual, see doc/APIchanges for more information.
-
-   Now let's talk about optimizations. Ronald S. Bultje made the VP9 decoder
-   usable on x86 32-bit systems and pre-ssse3 CPUs like Phenom (even dual core
-   Athlons can play 1080p 30fps VP9 content now), so we now secretly hope for
-   Google and Mozilla to use ffvp9 instead of libvpx. But VP9 is not the
-   center of attention anymore, and HEVC/H.265 is also getting many
-   improvements, which include C and x86 ASM optimizations, mainly from James
-   Almer, Christophe Gisquet and Pierre-Edouard Lepere.
-
-   Even though we had many x86 contributions, it is not the only architecture
-   getting some love, with Seppo Tomperi adding ARM NEON optimizations to the
-   HEVC stack, and James Cowgill adding MIPS64 assembly for all kind of audio
-   processing code in libavcodec.
-
-   And finally, Michael Niedermayer is still fixing many bugs, dealing with
-   most of the boring work such as making releases, applying tons of
-   contributors patches, and daily merging the changes from the Libav project.
-
-   A more complete Changelog is available at the root of the project, and the
-   complete Git history on http://source.ffmpeg.org.
-
-   We hope you will like this release as much as we enjoyed working on it, and
-   as usual, if you have any questions about it, or any FFmpeg related topic,
-   feel free to join us on the #ffmpeg IRC channel (on irc.freenode.net) or ask
-   on the mailing-lists.

VERSION

@@ -0,0 +1 @@
+2.1.4

arch.mak

@@ -1,14 +1,16 @@
 OBJS-$(HAVE_ARMV5TE) += $(ARMV5TE-OBJS) $(ARMV5TE-OBJS-yes)
 OBJS-$(HAVE_ARMV6)   += $(ARMV6-OBJS)   $(ARMV6-OBJS-yes)
-OBJS-$(HAVE_ARMV8)   += $(ARMV8-OBJS)   $(ARMV8-OBJS-yes)
 OBJS-$(HAVE_VFP)     += $(VFP-OBJS)     $(VFP-OBJS-yes)
 OBJS-$(HAVE_NEON)    += $(NEON-OBJS)    $(NEON-OBJS-yes)
 
 OBJS-$(HAVE_MIPSFPU)   += $(MIPSFPU-OBJS)    $(MIPSFPU-OBJS-yes)
+OBJS-$(HAVE_MIPS32R2)  += $(MIPS32R2-OBJS)   $(MIPS32R2-OBJS-yes)
 OBJS-$(HAVE_MIPSDSPR1) += $(MIPSDSPR1-OBJS)  $(MIPSDSPR1-OBJS-yes)
 OBJS-$(HAVE_MIPSDSPR2) += $(MIPSDSPR2-OBJS)  $(MIPSDSPR2-OBJS-yes)
 
 OBJS-$(HAVE_ALTIVEC) += $(ALTIVEC-OBJS) $(ALTIVEC-OBJS-yes)
 
+OBJS-$(HAVE_VIS)     += $(VIS-OBJS)     $(VIS-OBJS-yes)
+
 OBJS-$(HAVE_MMX)     += $(MMX-OBJS)     $(MMX-OBJS-yes)
 OBJS-$(HAVE_YASM)    += $(YASM-OBJS)    $(YASM-OBJS-yes)

cmdutils.c

@@ -20,7 +20,6 @@
  */
 
 #include <string.h>
-#include <stdint.h>
 #include <stdlib.h>
 #include <errno.h>
 #include <math.h>
@@ -49,8 +48,8 @@
 #include "libavutil/dict.h"
 #include "libavutil/opt.h"
 #include "libavutil/cpu.h"
-#include "libavutil/ffversion.h"
 #include "cmdutils.h"
+#include "version.h"
 #if CONFIG_NETWORK
 #include "libavformat/network.h"
 #endif
@@ -58,6 +57,10 @@
 #include <sys/time.h>
 #include <sys/resource.h>
 #endif
+#if CONFIG_OPENCL
+#include "libavutil/opencl.h"
+#endif
+
 
 static int init_report(const char *env);
 
@@ -65,9 +68,9 @@ struct SwsContext *sws_opts;
 AVDictionary *swr_opts;
 AVDictionary *format_opts, *codec_opts, *resample_opts;
 
+const int this_year = 2014;
+
 static FILE *report_file;
-static int report_file_level = AV_LOG_DEBUG;
-int hide_banner = 0;
 
 void init_opts(void)
 {
@@ -105,11 +108,9 @@ static void log_callback_report(void *ptr, int level, const char *fmt, va_list v
     av_log_default_callback(ptr, level, fmt, vl);
     av_log_format_line(ptr, level, fmt, vl2, line, sizeof(line), &print_prefix);
     va_end(vl2);
-    if (report_file_level >= level) {
     fputs(line, report_file);
     fflush(report_file);
 }
-}
 
 static void (*program_exit)(int ret);
 
@@ -166,7 +167,7 @@ void show_help_options(const OptionDef *options, const char *msg, int req_flags,
     int first;
 
     first = 1;
-    for (po = options; po->name; po++) {
+    for (po = options; po->name != NULL; po++) {
         char buf[64];
 
         if (((po->flags & req_flags) != req_flags) ||
@@ -205,7 +206,7 @@ static const OptionDef *find_option(const OptionDef *po, const char *name)
     const char *p = strchr(name, ':');
     int len = p ? p - name : strlen(name);
 
-    while (po->name) {
+    while (po->name != NULL) {
         if (!strncmp(name, po->name, len) && strlen(po->name) == len)
             break;
         po++;
@@ -254,7 +255,7 @@ static void prepare_app_arguments(int *argc_ptr, char ***argv_ptr)
 
     win32_argv_utf8 = av_mallocz(sizeof(char *) * (win32_argc + 1) + buffsize);
     argstr_flat     = (char *)win32_argv_utf8 + sizeof(char *) * (win32_argc + 1);
-    if (!win32_argv_utf8) {
+    if (win32_argv_utf8 == NULL) {
         LocalFree(argv_w);
         return;
     }
@@ -290,14 +291,10 @@ static int write_option(void *optctx, const OptionDef *po, const char *opt,
     if (po->flags & OPT_SPEC) {
         SpecifierOpt **so = dst;
         char *p = strchr(opt, ':');
-        char *str;
 
         dstcount = (int *)(so + 1);
         *so = grow_array(*so, sizeof(**so), dstcount, *dstcount + 1);
-        str = av_strdup(p ? p + 1 : "");
-        if (!str)
-            return AVERROR(ENOMEM);
-        (*so)[*dstcount - 1].specifier = str;
+        (*so)[*dstcount - 1].specifier = av_strdup(p ? p + 1 : "");
         dst = &(*so)[*dstcount - 1].u;
     }
 
@@ -305,8 +302,6 @@ static int write_option(void *optctx, const OptionDef *po, const char *opt,
         char *str;
         str = av_strdup(arg);
         av_freep(dst);
-        if (!str)
-            return AVERROR(ENOMEM);
         *(char **)dst = str;
     } else if (po->flags & OPT_BOOL || po->flags & OPT_INT) {
         *(int *)dst = parse_number_or_die(opt, arg, OPT_INT64, INT_MIN, INT_MAX);
@@ -450,7 +445,7 @@ int locate_option(int argc, char **argv, const OptionDef *options,
              (po->name && !strcmp(optname, po->name)))
             return i;
 
-        if (!po->name || po->flags & HAS_ARG)
+        if (po->flags & HAS_ARG)
             i++;
     }
     return 0;
@@ -501,9 +496,6 @@ void parse_loglevel(int argc, char **argv, const OptionDef *options)
             fflush(report_file);
         }
     }
-    idx = locate_option(argc, argv, options, "hide_banner");
-    if (idx)
-        hide_banner = 1;
 }
 
 static const AVOption *opt_find(void *obj, const char *name, const char *unit,
@@ -561,11 +553,6 @@ int opt_default(void *optctx, const char *opt, const char *arg)
         }
         consumed = 1;
     }
-#else
-    if (!consumed && !strcmp(opt, "sws_flags")) {
-        av_log(NULL, AV_LOG_WARNING, "Ignoring %s %s, due to disabled swscale\n", opt, arg);
-        consumed = 1;
-    }
 #endif
 #if CONFIG_SWRESAMPLE
     swr_class = swr_get_class();
@@ -676,7 +663,7 @@ static void init_parse_context(OptionParseContext *octx,
     memset(octx, 0, sizeof(*octx));
 
     octx->nb_groups = nb_groups;
-    octx->groups    = av_mallocz_array(octx->nb_groups, sizeof(*octx->groups));
+    octx->groups    = av_mallocz(sizeof(*octx->groups) * octx->nb_groups);
     if (!octx->groups)
         exit_program(1);
 
@@ -848,17 +835,10 @@ int opt_loglevel(void *optctx, const char *opt, const char *arg)
     };
     char *tail;
     int level;
-    int flags;
     int i;
 
-    flags = av_log_get_flags();
     tail = strstr(arg, "repeat");
-    if (tail)
-        flags &= ~AV_LOG_SKIP_REPEATED;
-    else
-        flags |= AV_LOG_SKIP_REPEATED;
-
-    av_log_set_flags(flags);
+    av_log_set_flags(tail ? 0 : AV_LOG_SKIP_REPEATED);
     if (tail == arg)
         arg += 6 + (arg[6]=='+');
     if(tail && !*arg)
@@ -940,13 +920,6 @@ static int init_report(const char *env)
             av_free(filename_template);
             filename_template = val;
             val = NULL;
-        } else if (!strcmp(key, "level")) {
-            char *tail;
-            report_file_level = strtol(val, &tail, 10);
-            if (*tail) {
-                av_log(NULL, AV_LOG_FATAL, "Invalid report file level\n");
-                exit_program(1);
-            }
         } else {
             av_log(NULL, AV_LOG_ERROR, "Unknown key '%s' in FFREPORT\n", key);
         }
@@ -965,10 +938,9 @@ static int init_report(const char *env)
 
     report_file = fopen(filename.str, "w");
     if (!report_file) {
-        int ret = AVERROR(errno);
         av_log(NULL, AV_LOG_ERROR, "Failed to open report \"%s\": %s\n",
                filename.str, strerror(errno));
-        return ret;
+        return AVERROR(errno);
     }
     av_log_set_callback(log_callback_report);
     av_log(NULL, AV_LOG_INFO,
@@ -1014,6 +986,26 @@ int opt_timelimit(void *optctx, const char *opt, const char *arg)
     return 0;
 }
 
+#if CONFIG_OPENCL
+int opt_opencl(void *optctx, const char *opt, const char *arg)
+{
+    char *key, *value;
+    const char *opts = arg;
+    int ret = 0;
+    while (*opts) {
+        ret = av_opt_get_key_value(&opts, "=", ":", 0, &key, &value);
+        if (ret < 0)
+            return ret;
+        ret = av_opencl_set_option(key, value);
+        if (ret < 0)
+            return ret;
+        if (*opts)
+            opts++;
+    }
+    return ret;
+}
+#endif
+
 void print_error(const char *filename, int err)
 {
     char errbuf[128];
@@ -1079,43 +1071,18 @@ static void print_program_info(int flags, int level)
     av_log(NULL, level, "%s version " FFMPEG_VERSION, program_name);
     if (flags & SHOW_COPYRIGHT)
         av_log(NULL, level, " Copyright (c) %d-%d the FFmpeg developers",
-               program_birth_year, CONFIG_THIS_YEAR);
+               program_birth_year, this_year);
     av_log(NULL, level, "\n");
-    av_log(NULL, level, "%sbuilt with %s\n", indent, CC_IDENT);
+    av_log(NULL, level, "%sbuilt on %s %s with %s\n",
+           indent, __DATE__, __TIME__, CC_IDENT);
 
     av_log(NULL, level, "%sconfiguration: " FFMPEG_CONFIGURATION "\n", indent);
 }
 
-static void print_buildconf(int flags, int level)
-{
-    const char *indent = flags & INDENT ? "  " : "";
-    char str[] = { FFMPEG_CONFIGURATION };
-    char *conflist, *remove_tilde, *splitconf;
-
-    // Change all the ' --' strings to '~--' so that
-    // they can be identified as tokens.
-    while ((conflist = strstr(str, " --")) != NULL) {
-        strncpy(conflist, "~--", 3);
-    }
-
-    // Compensate for the weirdness this would cause
-    // when passing 'pkg-config --static'.
-    while ((remove_tilde = strstr(str, "pkg-config~")) != NULL) {
-        strncpy(remove_tilde, "pkg-config ", 11);
-    }
-
-    splitconf = strtok(str, "~");
-    av_log(NULL, level, "\n%sconfiguration:\n", indent);
-    while (splitconf != NULL) {
-        av_log(NULL, level, "%s%s%s\n", indent, indent, splitconf);
-        splitconf = strtok(NULL, "~");
-    }
-}
-
 void show_banner(int argc, char **argv, const OptionDef *options)
 {
     int idx = locate_option(argc, argv, options, "version");
-    if (hide_banner || idx)
+    if (idx)
         return;
 
     print_program_info (INDENT|SHOW_COPYRIGHT, AV_LOG_INFO);
@@ -1126,20 +1093,12 @@ void show_banner(int argc, char **argv, const OptionDef *options)
 int show_version(void *optctx, const char *opt, const char *arg)
 {
     av_log_set_callback(log_callback_help);
-    print_program_info (SHOW_COPYRIGHT, AV_LOG_INFO);
+    print_program_info (0           , AV_LOG_INFO);
     print_all_libs_info(SHOW_VERSION, AV_LOG_INFO);
 
     return 0;
 }
 
-int show_buildconf(void *optctx, const char *opt, const char *arg)
-{
-    av_log_set_callback(log_callback_help);
-    print_buildconf      (INDENT|0, AV_LOG_INFO);
-
-    return 0;
-}
-
 int show_license(void *optctx, const char *opt, const char *arg)
 {
 #if CONFIG_NONFREE
@@ -1214,24 +1173,16 @@ int show_license(void *optctx, const char *opt, const char *arg)
     return 0;
 }
 
-static int is_device(const AVClass *avclass)
-{
-    if (!avclass)
-        return 0;
-    return AV_IS_INPUT_DEVICE(avclass->category) || AV_IS_OUTPUT_DEVICE(avclass->category);
-}
-
-static int show_formats_devices(void *optctx, const char *opt, const char *arg, int device_only)
+int show_formats(void *optctx, const char *opt, const char *arg)
 {
     AVInputFormat *ifmt  = NULL;
     AVOutputFormat *ofmt = NULL;
     const char *last_name;
-    int is_dev;
 
-    printf("%s\n"
+    printf("File formats:\n"
            " D. = Demuxing supported\n"
            " .E = Muxing supported\n"
-           " --\n", device_only ? "Devices:" : "File formats:");
+           " --\n");
     last_name = "000";
     for (;;) {
         int decode = 0;
@@ -1240,10 +1191,7 @@ static int show_formats_devices(void *optctx, const char *opt, const char *arg,
         const char *long_name = NULL;
 
         while ((ofmt = av_oformat_next(ofmt))) {
-            is_dev = is_device(ofmt->priv_class);
-            if (!is_dev && device_only)
-                continue;
-            if ((!name || strcmp(ofmt->name, name) < 0) &&
+            if ((name == NULL || strcmp(ofmt->name, name) < 0) &&
                 strcmp(ofmt->name, last_name) > 0) {
                 name      = ofmt->name;
                 long_name = ofmt->long_name;
@@ -1251,10 +1199,7 @@ static int show_formats_devices(void *optctx, const char *opt, const char *arg,
             }
         }
         while ((ifmt = av_iformat_next(ifmt))) {
-            is_dev = is_device(ifmt->priv_class);
-            if (!is_dev && device_only)
-                continue;
-            if ((!name || strcmp(ifmt->name, name) < 0) &&
+            if ((name == NULL || strcmp(ifmt->name, name) < 0) &&
                 strcmp(ifmt->name, last_name) > 0) {
                 name      = ifmt->name;
                 long_name = ifmt->long_name;
@@ -1263,7 +1208,7 @@ static int show_formats_devices(void *optctx, const char *opt, const char *arg,
             if (name && strcmp(ifmt->name, name) == 0)
                 decode = 1;
         }
-        if (!name)
+        if (name == NULL)
             break;
         last_name = name;
 
@@ -1276,16 +1221,6 @@ static int show_formats_devices(void *optctx, const char *opt, const char *arg,
     return 0;
 }
 
-int show_formats(void *optctx, const char *opt, const char *arg)
-{
-    return show_formats_devices(optctx, opt, arg, 0);
-}
-
-int show_devices(void *optctx, const char *opt, const char *arg)
-{
-    return show_formats_devices(optctx, opt, arg, 1);
-}
-
 #define PRINT_CODEC_SUPPORTED(codec, field, type, list_name, term, get_name) \
     if (codec->field) {                                                      \
         const type *p = codec->field;                                        \
@@ -1430,9 +1365,6 @@ int show_codecs(void *optctx, const char *opt, const char *arg)
         const AVCodecDescriptor *desc = codecs[i];
         const AVCodec *codec = NULL;
 
-        if (strstr(desc->name, "_deprecated"))
-            continue;
-
         printf(" ");
         printf(avcodec_find_decoder(desc->id) ? "D" : ".");
         printf(avcodec_find_encoder(desc->id) ? "E" : ".");
@@ -1544,8 +1476,7 @@ int show_protocols(void *optctx, const char *opt, const char *arg)
 
 int show_filters(void *optctx, const char *opt, const char *arg)
 {
-#if CONFIG_AVFILTER
-    const AVFilter *filter = NULL;
+    const AVFilter av_unused(*filter) = NULL;
     char descr[64], *descr_cur;
     int i, j;
     const AVFilterPad *pad;
@@ -1558,6 +1489,7 @@ int show_filters(void *optctx, const char *opt, const char *arg)
            "  V = Video input/output\n"
            "  N = Dynamic number and/or type of input/output\n"
            "  | = Source or sink filter\n");
+#if CONFIG_AVFILTER
     while ((filter = avfilter_next(filter))) {
         descr_cur = descr;
         for (i = 0; i < 2; i++) {
@@ -1582,13 +1514,11 @@ int show_filters(void *optctx, const char *opt, const char *arg)
                filter->process_command                        ? 'C' : '.',
                filter->name, descr, filter->description);
     }
-#else
-    printf("No filters available: libavfilter disabled\n");
 #endif
     return 0;
 }
 
-int show_colors(void *optctx, const char *opt, const char *arg)
+void show_colors(void *optctx, const char *opt, const char *arg)
 {
     const char *name;
     const uint8_t *rgb;
@@ -1598,8 +1528,6 @@ int show_colors(void *optctx, const char *opt, const char *arg)
 
     for (i = 0; name = av_get_known_color_name(i, &rgb); i++)
         printf("%-32s #%02x%02x%02x\n", name, rgb[0], rgb[1], rgb[2]);
-
-    return 0;
 }
 
 int show_pix_fmts(void *optctx, const char *opt, const char *arg)
@@ -1648,13 +1576,13 @@ int show_layouts(void *optctx, const char *opt, const char *arg)
         if (!name)
             continue;
         descr = av_get_channel_description((uint64_t)1 << i);
-        printf("%-14s %s\n", name, descr);
+        printf("%-12s%s\n", name, descr);
     }
     printf("\nStandard channel layouts:\n"
            "NAME        DECOMPOSITION\n");
     for (i = 0; !av_get_standard_channel_layout(i, &layout, &name); i++) {
         if (name) {
-            printf("%-14s ", name);
+            printf("%-12s", name);
             for (j = 1; j; j <<= 1)
                 if ((layout & j))
                     printf("%s%s", (layout & (j - 1)) ? "+" : "", av_get_channel_name(j));
@@ -1821,8 +1749,6 @@ int show_help(void *optctx, const char *opt, const char *arg)
     av_log_set_callback(log_callback_help);
 
     topic = av_strdup(arg ? arg : "");
-    if (!topic)
-        return AVERROR(ENOMEM);
     par = strchr(topic, '=');
     if (par)
         *par++ = 0;
@@ -1862,48 +1788,35 @@ int read_yesno(void)
 
 int cmdutils_read_file(const char *filename, char **bufptr, size_t *size)
 {
-    int64_t ret;
-    FILE *f = av_fopen_utf8(filename, "rb");
+    int ret;
+    FILE *f = fopen(filename, "rb");
 
     if (!f) {
-        ret = AVERROR(errno);
         av_log(NULL, AV_LOG_ERROR, "Cannot read file '%s': %s\n", filename,
                strerror(errno));
-        return ret;
+        return AVERROR(errno);
     }
-
-    ret = fseek(f, 0, SEEK_END);
-    if (ret == -1) {
-        ret = AVERROR(errno);
-        goto out;
+    fseek(f, 0, SEEK_END);
+    *size = ftell(f);
+    fseek(f, 0, SEEK_SET);
+    if (*size == (size_t)-1) {
+        av_log(NULL, AV_LOG_ERROR, "IO error: %s\n", strerror(errno));
+        fclose(f);
+        return AVERROR(errno);
     }
-
-    ret = ftell(f);
-    if (ret < 0) {
-        ret = AVERROR(errno);
-        goto out;
-    }
-    *size = ret;
-
-    ret = fseek(f, 0, SEEK_SET);
-    if (ret == -1) {
-        ret = AVERROR(errno);
-        goto out;
-    }
-
     *bufptr = av_malloc(*size + 1);
     if (!*bufptr) {
         av_log(NULL, AV_LOG_ERROR, "Could not allocate file buffer\n");
-        ret = AVERROR(ENOMEM);
-        goto out;
+        fclose(f);
+        return AVERROR(ENOMEM);
     }
     ret = fread(*bufptr, 1, *size, f);
     if (ret < *size) {
         av_free(*bufptr);
         if (ferror(f)) {
-            ret = AVERROR(errno);
             av_log(NULL, AV_LOG_ERROR, "Error while reading file '%s': %s\n",
                    filename, strerror(errno));
+            ret = AVERROR(errno);
         } else
             ret = AVERROR_EOF;
     } else {
@@ -1911,9 +1824,6 @@ int cmdutils_read_file(const char *filename, char **bufptr, size_t *size)
         (*bufptr)[(*size)++] = '\0';
     }
 
-out:
-    if (ret < 0)
-        av_log(NULL, AV_LOG_ERROR, "IO error: %s\n", av_err2str(ret));
     fclose(f);
     return ret;
 }
@@ -2013,12 +1923,11 @@ AVDictionary *filter_codec_opts(AVDictionary *opts, enum AVCodecID codec_id,
             switch (check_stream_specifier(s, st, p + 1)) {
             case  1: *p = 0; break;
             case  0:         continue;
-            default:         exit_program(1);
+            default:         return NULL;
             }
 
         if (av_opt_find(&cc, t->key, NULL, flags, AV_OPT_SEARCH_FAKE_OBJ) ||
-            !codec ||
-            (codec->priv_class &&
+            (codec && codec->priv_class &&
              av_opt_find(&codec->priv_class, t->key, NULL, flags,
                          AV_OPT_SEARCH_FAKE_OBJ)))
             av_dict_set(&ret, t->key, t->value, 0);
@@ -2041,7 +1950,7 @@ AVDictionary **setup_find_stream_info_opts(AVFormatContext *s,
 
     if (!s->nb_streams)
         return NULL;
-    opts = av_mallocz_array(s->nb_streams, sizeof(*opts));
+    opts = av_mallocz(s->nb_streams * sizeof(*opts));
     if (!opts) {
         av_log(NULL, AV_LOG_ERROR,
                "Could not alloc memory for stream options.\n");
@@ -2060,7 +1969,7 @@ void *grow_array(void *array, int elem_size, int *size, int new_size)
         exit_program(1);
     }
     if (*size < new_size) {
-        uint8_t *tmp = av_realloc_array(array, new_size, elem_size);
+        uint8_t *tmp = av_realloc(array, new_size*elem_size);
         if (!tmp) {
             av_log(NULL, AV_LOG_ERROR, "Could not alloc buffer.\n");
             exit_program(1);
@@ -2071,161 +1980,3 @@ void *grow_array(void *array, int elem_size, int *size, int new_size)
     }
     return array;
 }
-
-#if CONFIG_AVDEVICE
-static int print_device_sources(AVInputFormat *fmt, AVDictionary *opts)
-{
-    int ret, i;
-    AVDeviceInfoList *device_list = NULL;
-
-    if (!fmt || !fmt->priv_class  || !AV_IS_INPUT_DEVICE(fmt->priv_class->category))
-        return AVERROR(EINVAL);
-
-    printf("Audo-detected sources for %s:\n", fmt->name);
-    if (!fmt->get_device_list) {
-        ret = AVERROR(ENOSYS);
-        printf("Cannot list sources. Not implemented.\n");
-        goto fail;
-    }
-
-    if ((ret = avdevice_list_input_sources(fmt, NULL, opts, &device_list)) < 0) {
-        printf("Cannot list sources.\n");
-        goto fail;
-    }
-
-    for (i = 0; i < device_list->nb_devices; i++) {
-        printf("%s %s [%s]\n", device_list->default_device == i ? "*" : " ",
-               device_list->devices[i]->device_name, device_list->devices[i]->device_description);
-    }
-
-  fail:
-    avdevice_free_list_devices(&device_list);
-    return ret;
-}
-
-static int print_device_sinks(AVOutputFormat *fmt, AVDictionary *opts)
-{
-    int ret, i;
-    AVDeviceInfoList *device_list = NULL;
-
-    if (!fmt || !fmt->priv_class  || !AV_IS_OUTPUT_DEVICE(fmt->priv_class->category))
-        return AVERROR(EINVAL);
-
-    printf("Audo-detected sinks for %s:\n", fmt->name);
-    if (!fmt->get_device_list) {
-        ret = AVERROR(ENOSYS);
-        printf("Cannot list sinks. Not implemented.\n");
-        goto fail;
-    }
-
-    if ((ret = avdevice_list_output_sinks(fmt, NULL, opts, &device_list)) < 0) {
-        printf("Cannot list sinks.\n");
-        goto fail;
-    }
-
-    for (i = 0; i < device_list->nb_devices; i++) {
-        printf("%s %s [%s]\n", device_list->default_device == i ? "*" : " ",
-               device_list->devices[i]->device_name, device_list->devices[i]->device_description);
-    }
-
-  fail:
-    avdevice_free_list_devices(&device_list);
-    return ret;
-}
-
-static int show_sinks_sources_parse_arg(const char *arg, char **dev, AVDictionary **opts)
-{
-    int ret;
-    if (arg) {
-        char *opts_str = NULL;
-        av_assert0(dev && opts);
-        *dev = av_strdup(arg);
-        if (!*dev)
-            return AVERROR(ENOMEM);
-        if ((opts_str = strchr(*dev, ','))) {
-            *(opts_str++) = '\0';
-            if (opts_str[0] && ((ret = av_dict_parse_string(opts, opts_str, "=", ":", 0)) < 0)) {
-                av_freep(dev);
-                return ret;
-            }
-        }
-    } else
-        printf("\nDevice name is not provided.\n"
-                "You can pass devicename[,opt1=val1[,opt2=val2...]] as an argument.\n\n");
-    return 0;
-}
-
-int show_sources(void *optctx, const char *opt, const char *arg)
-{
-    AVInputFormat *fmt = NULL;
-    char *dev = NULL;
-    AVDictionary *opts = NULL;
-    int ret = 0;
-    int error_level = av_log_get_level();
-
-    av_log_set_level(AV_LOG_ERROR);
-
-    if ((ret = show_sinks_sources_parse_arg(arg, &dev, &opts)) < 0)
-        goto fail;
-
-    do {
-        fmt = av_input_audio_device_next(fmt);
-        if (fmt) {
-            if (!strcmp(fmt->name, "lavfi"))
-                continue; //it's pointless to probe lavfi
-            if (dev && !av_match_name(dev, fmt->name))
-                continue;
-            print_device_sources(fmt, opts);
-        }
-    } while (fmt);
-    do {
-        fmt = av_input_video_device_next(fmt);
-        if (fmt) {
-            if (dev && !av_match_name(dev, fmt->name))
-                continue;
-            print_device_sources(fmt, opts);
-        }
-    } while (fmt);
-  fail:
-    av_dict_free(&opts);
-    av_free(dev);
-    av_log_set_level(error_level);
-    return ret;
-}
-
-int show_sinks(void *optctx, const char *opt, const char *arg)
-{
-    AVOutputFormat *fmt = NULL;
-    char *dev = NULL;
-    AVDictionary *opts = NULL;
-    int ret = 0;
-    int error_level = av_log_get_level();
-
-    av_log_set_level(AV_LOG_ERROR);
-
-    if ((ret = show_sinks_sources_parse_arg(arg, &dev, &opts)) < 0)
-        goto fail;
-
-    do {
-        fmt = av_output_audio_device_next(fmt);
-        if (fmt) {
-            if (dev && !av_match_name(dev, fmt->name))
-                continue;
-            print_device_sinks(fmt, opts);
-        }
-    } while (fmt);
-    do {
-        fmt = av_output_video_device_next(fmt);
-        if (fmt) {
-            if (dev && !av_match_name(dev, fmt->name))
-                continue;
-            print_device_sinks(fmt, opts);
-        }
-    } while (fmt);
-  fail:
-    av_dict_free(&opts);
-    av_free(dev);
-    av_log_set_level(error_level);
-    return ret;
-}
-#endif

cmdutils.h

@@ -24,13 +24,12 @@
 
 #include <stdint.h>
 
-#include "config.h"
 #include "libavcodec/avcodec.h"
 #include "libavfilter/avfilter.h"
 #include "libavformat/avformat.h"
 #include "libswscale/swscale.h"
 
-#ifdef _WIN32
+#ifdef __MINGW32__
 #undef main /* We don't want SDL to override our main() */
 #endif
 
@@ -44,12 +43,16 @@ extern const char program_name[];
  */
 extern const int program_birth_year;
 
+/**
+ * this year, defined by the program for show_banner()
+ */
+extern const int this_year;
+
 extern AVCodecContext *avcodec_opts[AVMEDIA_TYPE_NB];
 extern AVFormatContext *avformat_opts;
 extern struct SwsContext *sws_opts;
 extern AVDictionary *swr_opts;
 extern AVDictionary *format_opts, *codec_opts, *resample_opts;
-extern int hide_banner;
 
 /**
  * Register a program-specific cleanup routine.
@@ -59,7 +62,7 @@ void register_exit(void (*cb)(int ret));
 /**
  * Wraps exit with a program-specific cleanup routine.
  */
-void exit_program(int ret) av_noreturn;
+void exit_program(int ret);
 
 /**
  * Initialize the cmdutils option system, in particular
@@ -100,12 +103,8 @@ int opt_max_alloc(void *optctx, const char *opt, const char *arg);
 
 int opt_codec_debug(void *optctx, const char *opt, const char *arg);
 
-#if CONFIG_OPENCL
 int opt_opencl(void *optctx, const char *opt, const char *arg);
 
-int opt_opencl_bench(void *optctx, const char *opt, const char *arg);
-#endif
-
 /**
  * Limit the execution time.
  */
@@ -415,13 +414,6 @@ void show_banner(int argc, char **argv, const OptionDef *options);
  */
 int show_version(void *optctx, const char *opt, const char *arg);
 
-/**
- * Print the build configuration of the program to stdout. The contents
- * depend on the definition of FFMPEG_CONFIGURATION.
- * This option processing function does not utilize the arguments.
- */
-int show_buildconf(void *optctx, const char *opt, const char *arg);
-
 /**
  * Print the license of the program to stdout. The license depends on
  * the license of the libraries compiled into the program.
@@ -431,31 +423,10 @@ int show_license(void *optctx, const char *opt, const char *arg);
 
 /**
  * Print a listing containing all the formats supported by the
- * program (including devices).
- * This option processing function does not utilize the arguments.
- */
-int show_formats(void *optctx, const char *opt, const char *arg);
-
-/**
- * Print a listing containing all the devices supported by the
  * program.
  * This option processing function does not utilize the arguments.
  */
-int show_devices(void *optctx, const char *opt, const char *arg);
-
-#if CONFIG_AVDEVICE
-/**
- * Print a listing containing audodetected sinks of the output device.
- * Device name with options may be passed as an argument to limit results.
- */
-int show_sinks(void *optctx, const char *opt, const char *arg);
-
-/**
- * Print a listing containing audodetected sources of the input device.
- * Device name with options may be passed as an argument to limit results.
- */
-int show_sources(void *optctx, const char *opt, const char *arg);
-#endif
+int show_formats(void *optctx, const char *opt, const char *arg);
 
 /**
  * Print a listing containing all the codecs supported by the
@@ -521,7 +492,7 @@ int show_sample_fmts(void *optctx, const char *opt, const char *arg);
  * Print a listing containing all the color names and values recognized
  * by the program.
  */
-int show_colors(void *optctx, const char *opt, const char *arg);
+void show_colors(void *optctx, const char *opt, const char *arg);
 
 /**
  * Return a positive value if a line read from standard input

cmdutils_common_opts.h

@@ -4,9 +4,7 @@
     { "help"       , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
     { "-help"      , OPT_EXIT, {.func_arg = show_help},         "show help", "topic" },
     { "version"    , OPT_EXIT, {.func_arg = show_version},      "show version" },
-    { "buildconf"  , OPT_EXIT, {.func_arg = show_buildconf},    "show build configuration" },
     { "formats"    , OPT_EXIT, {.func_arg = show_formats  },    "show available formats" },
-    { "devices"    , OPT_EXIT, {.func_arg = show_devices  },    "show available devices" },
     { "codecs"     , OPT_EXIT, {.func_arg = show_codecs   },    "show available codecs" },
     { "decoders"   , OPT_EXIT, {.func_arg = show_decoders },    "show available decoders" },
     { "encoders"   , OPT_EXIT, {.func_arg = show_encoders },    "show available encoders" },
@@ -22,14 +20,6 @@
     { "report"     , 0,        {(void*)opt_report}, "generate a report" },
     { "max_alloc"  , HAS_ARG,  {.func_arg = opt_max_alloc},     "set maximum size of a single allocated block", "bytes" },
     { "cpuflags"   , HAS_ARG | OPT_EXPERT, { .func_arg = opt_cpuflags }, "force specific cpu flags", "flags" },
-    { "hide_banner", OPT_BOOL | OPT_EXPERT, {&hide_banner},     "do not show program banner", "hide_banner" },
 #if CONFIG_OPENCL
-    { "opencl_bench", OPT_EXIT, {.func_arg = opt_opencl_bench}, "run benchmark on all OpenCL devices and show results" },
     { "opencl_options", HAS_ARG, {.func_arg = opt_opencl},      "set OpenCL environment options" },
 #endif
-#if CONFIG_AVDEVICE
-    { "sources"    , OPT_EXIT | HAS_ARG, { .func_arg = show_sources },
-      "list sources of the input device", "device" },
-    { "sinks"      , OPT_EXIT | HAS_ARG, { .func_arg = show_sinks },
-      "list sinks of the output device", "device" },
-#endif

cmdutils_opencl.c

@@ -1,276 +0,0 @@
-/*
- * Copyright (C) 2013 Lenny Wang
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include "libavutil/opt.h"
-#include "libavutil/time.h"
-#include "libavutil/log.h"
-#include "libavutil/opencl.h"
-#include "libavutil/avstring.h"
-#include "cmdutils.h"
-
-typedef struct {
-    int platform_idx;
-    int device_idx;
-    char device_name[64];
-    int64_t runtime;
-} OpenCLDeviceBenchmark;
-
-const char *ocl_bench_source = AV_OPENCL_KERNEL(
-inline unsigned char clip_uint8(int a)
-{
-    if (a & (~0xFF))
-        return (-a)>>31;
-    else
-        return a;
-}
-
-kernel void unsharp_bench(
-                    global unsigned char *src,
-                    global unsigned char *dst,
-                    global int *mask,
-                    int width,
-                    int height)
-{
-    int i, j, local_idx, lc_idx, sum = 0;
-    int2 thread_idx, block_idx, global_idx, lm_idx;
-    thread_idx.x = get_local_id(0);
-    thread_idx.y = get_local_id(1);
-    block_idx.x = get_group_id(0);
-    block_idx.y = get_group_id(1);
-    global_idx.x = get_global_id(0);
-    global_idx.y = get_global_id(1);
-    local uchar data[32][32];
-    local int lc[128];
-
-    for (i = 0; i <= 1; i++) {
-        lm_idx.y = -8 + (block_idx.y + i) * 16 + thread_idx.y;
-        lm_idx.y = lm_idx.y < 0 ? 0 : lm_idx.y;
-        lm_idx.y = lm_idx.y >= height ? height - 1: lm_idx.y;
-        for (j = 0; j <= 1; j++) {
-            lm_idx.x = -8 + (block_idx.x + j) * 16 + thread_idx.x;
-            lm_idx.x = lm_idx.x < 0 ? 0 : lm_idx.x;
-            lm_idx.x = lm_idx.x >= width ? width - 1: lm_idx.x;
-            data[i*16 + thread_idx.y][j*16 + thread_idx.x] = src[lm_idx.y*width + lm_idx.x];
-        }
-    }
-    local_idx = thread_idx.y*16 + thread_idx.x;
-    if (local_idx < 128)
-        lc[local_idx] = mask[local_idx];
-    barrier(CLK_LOCAL_MEM_FENCE);
-
-    \n#pragma unroll\n
-    for (i = -4; i <= 4; i++) {
-        lm_idx.y = 8 + i + thread_idx.y;
-        \n#pragma unroll\n
-        for (j = -4; j <= 4; j++) {
-            lm_idx.x = 8 + j + thread_idx.x;
-            lc_idx = (i + 4)*8 + j + 4;
-            sum += (int)data[lm_idx.y][lm_idx.x] * lc[lc_idx];
-        }
-    }
-    int temp = (int)data[thread_idx.y + 8][thread_idx.x + 8];
-    int res = temp + (((temp - (int)((sum + 1<<15) >> 16))) >> 16);
-    if (global_idx.x < width && global_idx.y < height)
-        dst[global_idx.x + global_idx.y*width] = clip_uint8(res);
-}
-);
-
-#define OCLCHECK(method, ... )                                                 \
-do {                                                                           \
-    status = method(__VA_ARGS__);                                              \
-    if (status != CL_SUCCESS) {                                                \
-        av_log(NULL, AV_LOG_ERROR, # method " error '%s'\n",                   \
-               av_opencl_errstr(status));                                      \
-        ret = AVERROR_EXTERNAL;                                                \
-        goto end;                                                              \
-    }                                                                          \
-} while (0)
-
-#define CREATEBUF(out, flags, size)                                            \
-do {                                                                           \
-    out = clCreateBuffer(ext_opencl_env->context, flags, size, NULL, &status); \
-    if (status != CL_SUCCESS) {                                                \
-        av_log(NULL, AV_LOG_ERROR, "Could not create OpenCL buffer\n");        \
-        ret = AVERROR_EXTERNAL;                                                \
-        goto end;                                                              \
-    }                                                                          \
-} while (0)
-
-static void fill_rand_int(int *data, int n)
-{
-    int i;
-    srand(av_gettime());
-    for (i = 0; i < n; i++)
-        data[i] = rand();
-}
-
-#define OPENCL_NB_ITER 5
-static int64_t run_opencl_bench(AVOpenCLExternalEnv *ext_opencl_env)
-{
-    int i, arg = 0, width = 1920, height = 1088;
-    int64_t start, ret = 0;
-    cl_int status;
-    size_t kernel_len;
-    char *inbuf;
-    int *mask;
-    int buf_size = width * height * sizeof(char);
-    int mask_size = sizeof(uint32_t) * 128;
-
-    cl_mem cl_mask, cl_inbuf, cl_outbuf;
-    cl_kernel kernel = NULL;
-    cl_program program = NULL;
-    size_t local_work_size_2d[2] = {16, 16};
-    size_t global_work_size_2d[2] = {(size_t)width, (size_t)height};
-
-    if (!(inbuf = av_malloc(buf_size)) || !(mask = av_malloc(mask_size))) {
-        av_log(NULL, AV_LOG_ERROR, "Out of memory\n");
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-    fill_rand_int((int*)inbuf, buf_size/4);
-    fill_rand_int(mask, mask_size/4);
-
-    CREATEBUF(cl_mask, CL_MEM_READ_ONLY, mask_size);
-    CREATEBUF(cl_inbuf, CL_MEM_READ_ONLY, buf_size);
-    CREATEBUF(cl_outbuf, CL_MEM_READ_WRITE, buf_size);
-
-    kernel_len = strlen(ocl_bench_source);
-    program = clCreateProgramWithSource(ext_opencl_env->context, 1, &ocl_bench_source,
-                                        &kernel_len, &status);
-    if (status != CL_SUCCESS || !program) {
-        av_log(NULL, AV_LOG_ERROR, "OpenCL unable to create benchmark program\n");
-        ret = AVERROR_EXTERNAL;
-        goto end;
-    }
-    status = clBuildProgram(program, 1, &(ext_opencl_env->device_id), NULL, NULL, NULL);
-    if (status != CL_SUCCESS) {
-        av_log(NULL, AV_LOG_ERROR, "OpenCL unable to build benchmark program\n");
-        ret = AVERROR_EXTERNAL;
-        goto end;
-    }
-    kernel = clCreateKernel(program, "unsharp_bench", &status);
-    if (status != CL_SUCCESS) {
-        av_log(NULL, AV_LOG_ERROR, "OpenCL unable to create benchmark kernel\n");
-        ret = AVERROR_EXTERNAL;
-        goto end;
-    }
-
-    OCLCHECK(clEnqueueWriteBuffer, ext_opencl_env->command_queue, cl_inbuf, CL_TRUE, 0,
-             buf_size, inbuf, 0, NULL, NULL);
-    OCLCHECK(clEnqueueWriteBuffer, ext_opencl_env->command_queue, cl_mask, CL_TRUE, 0,
-             mask_size, mask, 0, NULL, NULL);
-    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_mem), &cl_inbuf);
-    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_mem), &cl_outbuf);
-    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_mem), &cl_mask);
-    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_int), &width);
-    OCLCHECK(clSetKernelArg, kernel, arg++, sizeof(cl_int), &height);
-
-    start = av_gettime_relative();
-    for (i = 0; i < OPENCL_NB_ITER; i++)
-        OCLCHECK(clEnqueueNDRangeKernel, ext_opencl_env->command_queue, kernel, 2, NULL,
-                 global_work_size_2d, local_work_size_2d, 0, NULL, NULL);
-    clFinish(ext_opencl_env->command_queue);
-    ret = (av_gettime_relative() - start)/OPENCL_NB_ITER;
-end:
-    if (kernel)
-        clReleaseKernel(kernel);
-    if (program)
-        clReleaseProgram(program);
-    if (cl_inbuf)
-        clReleaseMemObject(cl_inbuf);
-    if (cl_outbuf)
-        clReleaseMemObject(cl_outbuf);
-    if (cl_mask)
-        clReleaseMemObject(cl_mask);
-    av_free(inbuf);
-    av_free(mask);
-    return ret;
-}
-
-static int compare_ocl_device_desc(const void *a, const void *b)
-{
-    return ((OpenCLDeviceBenchmark*)a)->runtime - ((OpenCLDeviceBenchmark*)b)->runtime;
-}
-
-int opt_opencl_bench(void *optctx, const char *opt, const char *arg)
-{
-    int i, j, nb_devices = 0, count = 0;
-    int64_t score = 0;
-    AVOpenCLDeviceList *device_list;
-    AVOpenCLDeviceNode *device_node = NULL;
-    OpenCLDeviceBenchmark *devices = NULL;
-    cl_platform_id platform;
-
-    av_opencl_get_device_list(&device_list);
-    for (i = 0; i < device_list->platform_num; i++)
-        nb_devices += device_list->platform_node[i]->device_num;
-    if (!nb_devices) {
-        av_log(NULL, AV_LOG_ERROR, "No OpenCL device detected!\n");
-        return AVERROR(EINVAL);
-    }
-    if (!(devices = av_malloc_array(nb_devices, sizeof(OpenCLDeviceBenchmark)))) {
-        av_log(NULL, AV_LOG_ERROR, "Could not allocate buffer\n");
-        return AVERROR(ENOMEM);
-    }
-
-    for (i = 0; i < device_list->platform_num; i++) {
-        for (j = 0; j < device_list->platform_node[i]->device_num; j++) {
-            device_node = device_list->platform_node[i]->device_node[j];
-            platform = device_list->platform_node[i]->platform_id;
-            score = av_opencl_benchmark(device_node, platform, run_opencl_bench);
-            if (score > 0) {
-                devices[count].platform_idx = i;
-                devices[count].device_idx = j;
-                devices[count].runtime = score;
-                av_strlcpy(devices[count].device_name, device_node->device_name,
-                           sizeof(devices[count].device_name));
-                count++;
-            }
-        }
-    }
-    qsort(devices, count, sizeof(OpenCLDeviceBenchmark), compare_ocl_device_desc);
-    fprintf(stderr, "platform_idx\tdevice_idx\tdevice_name\truntime\n");
-    for (i = 0; i < count; i++)
-        fprintf(stdout, "%d\t%d\t%s\t%"PRId64"\n",
-                devices[i].platform_idx, devices[i].device_idx,
-                devices[i].device_name, devices[i].runtime);
-
-    av_opencl_free_device_list(&device_list);
-    av_free(devices);
-    return 0;
-}
-
-int opt_opencl(void *optctx, const char *opt, const char *arg)
-{
-    char *key, *value;
-    const char *opts = arg;
-    int ret = 0;
-    while (*opts) {
-        ret = av_opt_get_key_value(&opts, "=", ":", 0, &key, &value);
-        if (ret < 0)
-            return ret;
-        ret = av_opencl_set_option(key, value);
-        if (ret < 0)
-            return ret;
-        if (*opts)
-            opts++;
-    }
-    return ret;
-}

common.mak

@@ -5,20 +5,12 @@
 # first so "all" becomes default target
 all: all-yes
 
-DEFAULT_YASMD=.dbg
-
-ifeq (1, DBG)
-YASMD=$(DEFAULT_YASMD)
-else
-YASMD=
-endif
-
 ifndef SUBDIR
 
 ifndef V
 Q      = @
 ECHO   = printf "$(1)\t%s\n" $(2)
-BRIEF  = CC CXX HOSTCC HOSTLD AS YASM AR LD STRIP CP WINDRES
+BRIEF  = CC CXX HOSTCC HOSTLD AS YASM AR LD STRIP CP
 SILENT = DEPCC DEPHOSTCC DEPAS DEPYASM RANLIB RM
 
 MSG    = $@
@@ -51,7 +43,6 @@ endef
 COMPILE_C = $(call COMPILE,CC)
 COMPILE_CXX = $(call COMPILE,CXX)
 COMPILE_S = $(call COMPILE,AS)
-COMPILE_HOSTC = $(call COMPILE,HOSTCC)
 
 %.o: %.c
 	$(COMPILE_C)
@@ -59,21 +50,12 @@ COMPILE_HOSTC = $(call COMPILE,HOSTCC)
 %.o: %.cpp
 	$(COMPILE_CXX)
 
-%.o: %.m
-	$(COMPILE_C)
-
 %.s: %.c
 	$(CC) $(CPPFLAGS) $(CFLAGS) -S -o $@ $<
 
 %.o: %.S
 	$(COMPILE_S)
 
-%_host.o: %.c
-	$(COMPILE_HOSTC)
-
-%.o: %.rc
-	$(WINDRES) $(IFLAGS) --preprocessor "$(DEPWINDRES) -E -xc-header -DRC_INVOKED $(CC_DEPFLAGS)" -o $@ $<
-
 %.i: %.c
 	$(CC) $(CCFLAGS) $(CC_E) $<
 
@@ -100,15 +82,14 @@ endif
 include $(SRC_PATH)/arch.mak
 
 OBJS      += $(OBJS-yes)
-SLIBOBJS  += $(SLIBOBJS-yes)
-FFLIBS    := $($(NAME)_FFLIBS) $(FFLIBS-yes) $(FFLIBS)
+FFLIBS    := $(FFLIBS-yes) $(FFLIBS)
 TESTPROGS += $(TESTPROGS-yes)
 
 LDLIBS       = $(FFLIBS:%=%$(BUILDSUF))
 FFEXTRALIBS := $(LDLIBS:%=$(LD_LIB)) $(EXTRALIBS)
 
+EXAMPLES  := $(EXAMPLES:%=$(SUBDIR)%-example$(EXESUF))
 OBJS      := $(sort $(OBJS:%=$(SUBDIR)%))
-SLIBOBJS  := $(sort $(SLIBOBJS:%=$(SUBDIR)%))
 TESTOBJS  := $(TESTOBJS:%=$(SUBDIR)%) $(TESTPROGS:%=$(SUBDIR)%-test.o)
 TESTPROGS := $(TESTPROGS:%=$(SUBDIR)%-test$(EXESUF))
 HOSTOBJS  := $(HOSTPROGS:%=$(SUBDIR)%.o)
@@ -132,31 +113,30 @@ checkheaders: $(HOBJS)
 alltools: $(TOOLS)
 
 $(HOSTOBJS): %.o: %.c
-	$(COMPILE_HOSTC)
+	$(call COMPILE,HOSTCC)
 
 $(HOSTPROGS): %$(HOSTEXESUF): %.o
-	$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $^ $(HOSTLIBS)
+	$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $< $(HOSTLIBS)
 
 $(OBJS):     | $(sort $(dir $(OBJS)))
 $(HOBJS):    | $(sort $(dir $(HOBJS)))
 $(HOSTOBJS): | $(sort $(dir $(HOSTOBJS)))
-$(SLIBOBJS): | $(sort $(dir $(SLIBOBJS)))
 $(TESTOBJS): | $(sort $(dir $(TESTOBJS)))
 $(TOOLOBJS): | tools
 
-OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(SLIBOBJS) $(TESTOBJS))
+OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(TESTOBJS))
 
-CLEANSUFFIXES     = *.d *.o *~ *.h.c *.map *.ver *.ho *.gcno *.gcda *$(DEFAULT_YASMD).asm
+CLEANSUFFIXES     = *.d *.o *~ *.h.c *.map *.ver *.ho *.gcno *.gcda
 DISTCLEANSUFFIXES = *.pc
 LIBSUFFIXES       = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a
 
 define RULES
 clean::
-	$(RM) $(OBJS) $(OBJS:.o=.d) $(OBJS:.o=$(DEFAULT_YASMD).d)
+	$(RM) $(OBJS) $(OBJS:.o=.d)
 	$(RM) $(HOSTPROGS)
 	$(RM) $(TOOLS)
 endef
 
 $(eval $(RULES))
 
--include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SLIBOBJS:.o=.d)) $(OBJS:.o=$(DEFAULT_YASMD).d)
+-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d))

compat/avisynth/avisynth_c.h

@@ -13,8 +13,7 @@
 //
 // You should have received a copy of the GNU General Public License
 // along with this program; if not, write to the Free Software
-// Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston,
-// MA 02110-1301 USA, or visit
+// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
 // http://www.gnu.org/copyleft/gpl.html .
 //
 // As a special exception, I give you permission to link to the
@@ -805,7 +804,7 @@ struct AVS_Library {
 
 AVSC_INLINE AVS_Library * avs_load_library() {
   AVS_Library *library = (AVS_Library *)malloc(sizeof(AVS_Library));
-  if (!library)
+  if (library == NULL)
     return NULL;
   library->handle = LoadLibrary("avisynth");
   if (library->handle == NULL)
@@ -870,7 +869,7 @@ fail:
 }
 
 AVSC_INLINE void avs_free_library(AVS_Library *library) {
-  if (!library)
+  if (library == NULL)
     return;
   FreeLibrary(library->handle);
   free(library);

compat/avisynth/avxsynth_c.h

@@ -13,8 +13,7 @@
 //
 // You should have received a copy of the GNU General Public License
 // along with this program; if not, write to the Free Software
-// Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston,
-// MA 02110-1301 USA, or visit
+// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
 // http://www.gnu.org/copyleft/gpl.html .
 //
 // As a special exception, I give you permission to link to the
@@ -513,21 +512,21 @@ AVSC_INLINE AVS_Value avs_array_elt(AVS_Value v, int index)
 // only use these functions on am AVS_Value that does not already have
 // an active value.  Remember, treat AVS_Value as a fat pointer.
 AVSC_INLINE AVS_Value avs_new_value_bool(int v0)
-        { AVS_Value v = {0}; v.type = 'b'; v.d.boolean = v0 == 0 ? 0 : 1; return v; }
+        { AVS_Value v; v.type = 'b'; v.d.boolean = v0 == 0 ? 0 : 1; return v; }
 AVSC_INLINE AVS_Value avs_new_value_int(int v0)
-        { AVS_Value v = {0}; v.type = 'i'; v.d.integer = v0; return v; }
+        { AVS_Value v; v.type = 'i'; v.d.integer = v0; return v; }
 AVSC_INLINE AVS_Value avs_new_value_string(const char * v0)
-        { AVS_Value v = {0}; v.type = 's'; v.d.string = v0; return v; }
+        { AVS_Value v; v.type = 's'; v.d.string = v0; return v; }
 AVSC_INLINE AVS_Value avs_new_value_float(float v0)
-        { AVS_Value v = {0}; v.type = 'f'; v.d.floating_pt = v0; return v;}
+        { AVS_Value v; v.type = 'f'; v.d.floating_pt = v0; return v;}
 AVSC_INLINE AVS_Value avs_new_value_error(const char * v0)
-        { AVS_Value v = {0}; v.type = 'e'; v.d.string = v0; return v; }
+        { AVS_Value v; v.type = 'e'; v.d.string = v0; return v; }
 #ifndef AVSC_NO_DECLSPEC
 AVSC_INLINE AVS_Value avs_new_value_clip(AVS_Clip * v0)
-        { AVS_Value v = {0}; avs_set_to_clip(&v, v0); return v; }
+        { AVS_Value v; avs_set_to_clip(&v, v0); return v; }
 #endif
 AVSC_INLINE AVS_Value avs_new_value_array(AVS_Value * v0, int size)
-        { AVS_Value v = {0}; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }
+        { AVS_Value v; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }
 
 /////////////////////////////////////////////////////////////////////
 //

compat/avisynth/windowsPorts/windows2linux.h

@@ -52,8 +52,8 @@ namespace avxsynth {
 //
 // Functions
 //
-#define MAKEDWORD(a,b,c,d) (((a) << 24) | ((b) << 16) | ((c) << 8) | (d))
-#define MAKEWORD(a,b) (((a) << 8) | (b))
+#define MAKEDWORD(a,b,c,d) ((a << 24) | (b << 16) | (c << 8) | (d))
+#define MAKEWORD(a,b) ((a << 8) | (b))
 
 #define lstrlen                             strlen
 #define lstrcpy                             strcpy

compat/float/float.h

@@ -1,35 +0,0 @@
-/*
- * Work around broken floating point limits on some systems.
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include_next <float.h>
-
-#ifdef FLT_MAX
-#undef  FLT_MAX
-#define FLT_MAX 3.40282346638528859812e+38F
-
-#undef  FLT_MIN
-#define FLT_MIN 1.17549435082228750797e-38F
-
-#undef  DBL_MAX
-#define DBL_MAX ((double)1.79769313486231570815e+308L)
-
-#undef  DBL_MIN
-#define DBL_MIN ((double)2.22507385850720138309e-308L)
-#endif

compat/float/limits.h

@@ -1,22 +0,0 @@
-/*
- * Work around broken floating point limits on some systems.
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include_next <limits.h>
-#include <float.h>

compat/getopt.c

@@ -38,6 +38,8 @@ static int optind = 1;
 static int optopt;
 static char *optarg;
 
+#undef fprintf
+
 static int getopt(int argc, char *argv[], char *opts)
 {
     static int sp = 1;
@@ -54,7 +56,7 @@ static int getopt(int argc, char *argv[], char *opts)
         }
     }
     optopt = c = argv[optind][sp];
-    if (c == ':' || !(cp = strchr(opts, c))) {
+    if (c == ':' || (cp = strchr(opts, c)) == NULL) {
         fprintf(stderr, ": illegal option -- %c\n", c);
         if (argv[optind][++sp] == '\0') {
             optind++;

compat/os2threads.h

@@ -32,8 +32,6 @@
 #undef __STRICT_ANSI__          /* for _beginthread() */
 #include <stdlib.h>
 
-#include "libavutil/mem.h"
-
 typedef TID  pthread_t;
 typedef void pthread_attr_t;
 

compat/va_copy.h

@@ -24,6 +24,3 @@
 #if !defined(va_copy) && defined(_MSC_VER)
 #define va_copy(dst, src) ((dst) = (src))
 #endif
-#if !defined(va_copy) && defined(__GNUC__) && __GNUC__ < 3
-#define va_copy(dst, src) __va_copy(dst, src)
-#endif

compat/w32pthreads.h

@@ -39,7 +39,6 @@
 #include <windows.h>
 #include <process.h>
 
-#include "libavutil/attributes.h"
 #include "libavutil/common.h"
 #include "libavutil/internal.h"
 #include "libavutil/mem.h"
@@ -55,29 +54,35 @@ typedef struct pthread_t {
  * not mutexes */
 typedef CRITICAL_SECTION pthread_mutex_t;
 
-/* This is the CONDITION_VARIABLE typedef for using Windows' native
- * conditional variables on kernels 6.0+. */
-#if HAVE_CONDITION_VARIABLE_PTR
-typedef CONDITION_VARIABLE pthread_cond_t;
-#else
+/* This is the CONDITIONAL_VARIABLE typedef for using Window's native
+ * conditional variables on kernels 6.0+.
+ * MinGW does not currently have this typedef. */
 typedef struct pthread_cond_t {
-    void *Ptr;
+    void *ptr;
 } pthread_cond_t;
+
+/* function pointers to conditional variable API on windows 6.0+ kernels */
+#if _WIN32_WINNT < 0x0600
+static void (WINAPI *cond_broadcast)(pthread_cond_t *cond);
+static void (WINAPI *cond_init)(pthread_cond_t *cond);
+static void (WINAPI *cond_signal)(pthread_cond_t *cond);
+static BOOL (WINAPI *cond_wait)(pthread_cond_t *cond, pthread_mutex_t *mutex,
+                                DWORD milliseconds);
+#else
+#define cond_init      InitializeConditionVariable
+#define cond_broadcast WakeAllConditionVariable
+#define cond_signal    WakeConditionVariable
+#define cond_wait      SleepConditionVariableCS
 #endif
 
-#if _WIN32_WINNT >= 0x0600
-#define InitializeCriticalSection(x) InitializeCriticalSectionEx(x, 0, 0)
-#define WaitForSingleObject(a, b) WaitForSingleObjectEx(a, b, FALSE)
-#endif
-
-static av_unused unsigned __stdcall attribute_align_arg win32thread_worker(void *arg)
+static unsigned __stdcall attribute_align_arg win32thread_worker(void *arg)
 {
     pthread_t *h = arg;
     h->ret = h->func(h->arg);
     return 0;
 }
 
-static av_unused int pthread_create(pthread_t *thread, const void *unused_attr,
+static int pthread_create(pthread_t *thread, const void *unused_attr,
                           void *(*start_routine)(void*), void *arg)
 {
     thread->func   = start_routine;
@@ -87,7 +92,7 @@ static av_unused int pthread_create(pthread_t *thread, const void *unused_attr,
     return !thread->handle;
 }
 
-static av_unused void pthread_join(pthread_t thread, void **value_ptr)
+static void pthread_join(pthread_t thread, void **value_ptr)
 {
     DWORD ret = WaitForSingleObject(thread.handle, INFINITE);
     if (ret != WAIT_OBJECT_0)
@@ -118,36 +123,6 @@ static inline int pthread_mutex_unlock(pthread_mutex_t *m)
     return 0;
 }
 
-#if _WIN32_WINNT >= 0x0600
-static inline int pthread_cond_init(pthread_cond_t *cond, const void *unused_attr)
-{
-    InitializeConditionVariable(cond);
-    return 0;
-}
-
-/* native condition variables do not destroy */
-static inline void pthread_cond_destroy(pthread_cond_t *cond)
-{
-    return;
-}
-
-static inline void pthread_cond_broadcast(pthread_cond_t *cond)
-{
-    WakeAllConditionVariable(cond);
-}
-
-static inline int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
-{
-    SleepConditionVariableCS(cond, mutex, INFINITE);
-    return 0;
-}
-
-static inline void pthread_cond_signal(pthread_cond_t *cond)
-{
-    WakeConditionVariable(cond);
-}
-
-#else // _WIN32_WINNT < 0x0600
 /* for pre-Windows 6.0 platforms we need to define and use our own condition
  * variable and api */
 typedef struct  win32_cond_t {
@@ -159,41 +134,33 @@ typedef struct  win32_cond_t {
     volatile int is_broadcast;
 } win32_cond_t;
 
-/* function pointers to conditional variable API on windows 6.0+ kernels */
-static void (WINAPI *cond_broadcast)(pthread_cond_t *cond);
-static void (WINAPI *cond_init)(pthread_cond_t *cond);
-static void (WINAPI *cond_signal)(pthread_cond_t *cond);
-static BOOL (WINAPI *cond_wait)(pthread_cond_t *cond, pthread_mutex_t *mutex,
-                                DWORD milliseconds);
-
-static av_unused int pthread_cond_init(pthread_cond_t *cond, const void *unused_attr)
+static void pthread_cond_init(pthread_cond_t *cond, const void *unused_attr)
 {
     win32_cond_t *win32_cond = NULL;
     if (cond_init) {
         cond_init(cond);
-        return 0;
+        return;
     }
 
     /* non native condition variables */
     win32_cond = av_mallocz(sizeof(win32_cond_t));
     if (!win32_cond)
-        return ENOMEM;
-    cond->Ptr = win32_cond;
+        return;
+    cond->ptr = win32_cond;
     win32_cond->semaphore = CreateSemaphore(NULL, 0, 0x7fffffff, NULL);
     if (!win32_cond->semaphore)
-        return ENOMEM;
+        return;
     win32_cond->waiters_done = CreateEvent(NULL, TRUE, FALSE, NULL);
     if (!win32_cond->waiters_done)
-        return ENOMEM;
+        return;
 
     pthread_mutex_init(&win32_cond->mtx_waiter_count, NULL);
     pthread_mutex_init(&win32_cond->mtx_broadcast, NULL);
-    return 0;
 }
 
-static av_unused void pthread_cond_destroy(pthread_cond_t *cond)
+static void pthread_cond_destroy(pthread_cond_t *cond)
 {
-    win32_cond_t *win32_cond = cond->Ptr;
+    win32_cond_t *win32_cond = cond->ptr;
     /* native condition variables do not destroy */
     if (cond_init)
         return;
@@ -204,12 +171,12 @@ static av_unused void pthread_cond_destroy(pthread_cond_t *cond)
     pthread_mutex_destroy(&win32_cond->mtx_waiter_count);
     pthread_mutex_destroy(&win32_cond->mtx_broadcast);
     av_freep(&win32_cond);
-    cond->Ptr = NULL;
+    cond->ptr = NULL;
 }
 
-static av_unused void pthread_cond_broadcast(pthread_cond_t *cond)
+static void pthread_cond_broadcast(pthread_cond_t *cond)
 {
-    win32_cond_t *win32_cond = cond->Ptr;
+    win32_cond_t *win32_cond = cond->ptr;
     int have_waiter;
 
     if (cond_broadcast) {
@@ -238,9 +205,9 @@ static av_unused void pthread_cond_broadcast(pthread_cond_t *cond)
     pthread_mutex_unlock(&win32_cond->mtx_broadcast);
 }
 
-static av_unused int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
+static int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
 {
-    win32_cond_t *win32_cond = cond->Ptr;
+    win32_cond_t *win32_cond = cond->ptr;
     int last_waiter;
     if (cond_wait) {
         cond_wait(cond, mutex, INFINITE);
@@ -270,9 +237,9 @@ static av_unused int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mu
     return pthread_mutex_lock(mutex);
 }
 
-static av_unused void pthread_cond_signal(pthread_cond_t *cond)
+static void pthread_cond_signal(pthread_cond_t *cond)
 {
-    win32_cond_t *win32_cond = cond->Ptr;
+    win32_cond_t *win32_cond = cond->ptr;
     int have_waiter;
     if (cond_signal) {
         cond_signal(cond);
@@ -294,9 +261,8 @@ static av_unused void pthread_cond_signal(pthread_cond_t *cond)
 
     pthread_mutex_unlock(&win32_cond->mtx_broadcast);
 }
-#endif
 
-static av_unused void w32thread_init(void)
+static void w32thread_init(void)
 {
 #if _WIN32_WINNT < 0x0600
     HANDLE kernel_dll = GetModuleHandle(TEXT("kernel32.dll"));

compat/windows/makedef

@@ -1,132 +0,0 @@
-#!/bin/sh
-
-# Copyright (c) 2013, Derek Buitenhuis
-#
-# Permission to use, copy, modify, and/or distribute this software for any
-# purpose with or without fee is hereby granted, provided that the above
-# copyright notice and this permission notice appear in all copies.
-#
-# THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
-# WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
-# MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
-# ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
-# WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
-# ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
-# OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
-
-# mktemp isn't POSIX, so supply an implementation
-mktemp() {
-    echo "${2%%XXX*}.${HOSTNAME}.${UID}.$$"
-}
-
-if [ $# -lt 2 ]; then
-    echo "Usage: makedef <version_script> <objects>" >&2
-    exit 0
-fi
-
-vscript=$1
-shift
-
-if [ ! -f "$vscript" ]; then
-    echo "Version script does not exist" >&2
-    exit 1
-fi
-
-for object in "$@"; do
-    if [ ! -f "$object" ]; then
-        echo "Object does not exist: ${object}" >&2
-        exit 1
-    fi
-done
-
-# Create a lib temporarily to dump symbols from.
-# It's just much easier to do it this way
-libname=$(mktemp -u "library").lib
-
-trap 'rm -f -- $libname' EXIT
-
-lib -out:${libname} $@ >/dev/null
-if [ $? != 0 ]; then
-    echo "Could not create temporary library." >&2
-    exit 1
-fi
-
-IFS='
-'
-
-# Determine if we're building for x86 or x86_64 and
-# set the symbol prefix accordingly.
-prefix=""
-arch=$(dumpbin -headers ${libname} |
-       tr '\t' ' ' |
-       grep '^ \+.\+machine \+(.\+)' |
-       head -1 |
-       sed -e 's/^ \{1,\}.\{1,\} \{1,\}machine \{1,\}(\(...\)).*/\1/')
-
-if [ "${arch}" = "x86" ]; then
-    prefix="_"
-else
-    if [ "${arch}" != "ARM" ] && [ "${arch}" != "x64" ]; then
-        echo "Unknown machine type." >&2
-        exit 1
-    fi
-fi
-
-started=0
-regex="none"
-
-for line in $(cat ${vscript} | tr '\t' ' '); do
-    # We only care about global symbols
-    echo "${line}" | grep -q '^ \+global:'
-    if [ $? = 0 ]; then
-        started=1
-        line=$(echo "${line}" | sed -e 's/^ \{1,\}global: *//')
-    else
-        echo "${line}" | grep -q '^ \+local:'
-        if [ $? = 0 ]; then
-            started=0
-        fi
-    fi
-
-    if [ ${started} = 0 ]; then
-        continue
-    fi
-
-    # Handle multiple symbols on one line
-    IFS=';'
-
-    # Work around stupid expansion to filenames
-    line=$(echo "${line}" | sed -e 's/\*/.\\+/g')
-    for exp in ${line}; do
-        # Remove leading and trailing whitespace
-        exp=$(echo "${exp}" | sed -e 's/^ *//' -e 's/ *$//')
-
-        if [ "${regex}" = "none" ]; then
-            regex="${exp}"
-        else
-            regex="${regex};${exp}"
-        fi
-    done
-
-    IFS='
-'
-done
-
-dump=$(dumpbin -linkermember:1 ${libname})
-
-rm ${libname}
-
-IFS=';'
-list=""
-for exp in ${regex}; do
-    list="${list}"'
-'$(echo "${dump}" |
-          sed -e '/public symbols/,$!d' -e '/^ \{1,\}Summary/,$d' -e "s/ \{1,\}${prefix}/ /" -e 's/ \{1,\}/ /g' |
-          tail -n +2 |
-          cut -d' ' -f3 |
-          grep "^${exp}" |
-          sed -e 's/^/    /')
-done
-
-echo "EXPORTS"
-echo "${list}" | sort | uniq | tail -n +2

doc/APIchanges

@@ -2,573 +2,44 @@ Never assume the API of libav* to be stable unless at least 1 month has passed
 since the last major version increase or the API was added.
 
 The last version increases were:
-libavcodec:    2014-08-09
-libavdevice:   2014-08-09
-libavfilter:   2014-08-09
-libavformat:   2014-08-09
-libavresample: 2014-08-09
-libpostproc:   2014-08-09
-libswresample: 2014-08-09
-libswscale:    2014-08-09
-libavutil:     2014-08-09
+libavcodec:    2013-03-xx
+libavdevice:   2013-03-xx
+libavfilter:   2012-06-22
+libavformat:   2013-03-xx
+libavresample: 2012-10-05
+libpostproc:   2011-04-18
+libswresample: 2011-09-19
+libswscale:    2011-06-20
+libavutil:     2012-10-22
 
 
 API changes, most recent first:
 
--------- 8< --------- FFmpeg 2.6 was cut here -------- 8< ---------
-
-2015-03-04 - cca4476 - lavf 56.25.100
-  Add avformat_flush()
-
-2015-03-03 - 81a9126 - lavf 56.24.100
-  Add avio_put_str16be()
-
-2015-02-19 - 560eb71 / 31d2039 - lavc 56.23.100 / 56.13.0
-  Add width, height, coded_width, coded_height and format to
-  AVCodecParserContext.
-
-2015-02-19 - e375511 / 5b1d9ce - lavu 54.19.100 / 54.9.0
-  Add AV_PIX_FMT_QSV for QSV hardware acceleration.
-
-2015-02-14 - ba22295 - lavc 56.21.102
-  Deprecate VIMA decoder.
-
-2015-01-27 - 62a82c6 / 728685f - lavc 56.21.100 / 56.12.0, lavu 54.18.100 / 54.8.0 - avcodec.h, frame.h
-  Add AV_PKT_DATA_AUDIO_SERVICE_TYPE and AV_FRAME_DATA_AUDIO_SERVICE_TYPE for
-  storing the audio service type as side data.
-
-2015-01-16 - a47c933 - lavf 56.19.100 - avformat.h
-  Add data_codec and data_codec_id for storing codec of data stream
-
-2015-01-11 - 007c33d - lavd 56.4.100 - avdevice.h
-  Add avdevice_list_input_sources().
-  Add avdevice_list_output_sinks().
-
-2014-12-25 - d7aaeea / c220a60 - lavc 56.19.100 / 56.10.0 - vdpau.h
-  Add av_vdpau_get_surface_parameters().
-
-2014-12-25 - ddb9a24 / 6c99c92 - lavc 56.18.100 / 56.9.0 - avcodec.h
-  Add AV_HWACCEL_FLAG_ALLOW_HIGH_DEPTH flag to av_vdpau_bind_context().
-
-2014-12-25 - d16079a / 57b6704 - lavc 56.17.100 / 56.8.0 - avcodec.h
-  Add AVCodecContext.sw_pix_fmt.
-
-2014-12-04 - 6e9ac02 - lavc 56.14.100 - dv_profile.h
-  Add av_dv_codec_profile2().
-
--------- 8< --------- FFmpeg 2.5 was cut here -------- 8< ---------
-
-2014-11-21 - ab922f9 - lavu 54.15.100 - dict.h
-   Add av_dict_get_string().
-
-2014-11-18 - a54a51c - lavu 54.14.100 - float_dsp.h
-  Add avpriv_float_dsp_alloc().
-
-2014-11-16 - 6690d4c3 - lavf 56.13.100 - avformat.h
-  Add AVStream.recommended_encoder_configuration with accessors.
-
-2014-11-16 - bee5844d - lavu 54.13.100 - opt.h
-  Add av_opt_serialize().
-
-2014-11-16 - eec69332 - lavu 54.12.100 - opt.h
-  Add av_opt_is_set_to_default().
-
-2014-11-06 - 44fa267 / 5e80fb7 - lavc 56.11.100 / 56.6.0 - vorbis_parser.h
-  Add a public API for parsing vorbis packets.
-
-2014-10-15 - 17085a0 / 7ea1b34 - lavc 56.7.100 / 56.5.0 - avcodec.h
-  Replace AVCodecContext.time_base used for decoding
-  with AVCodecContext.framerate.
-
-2014-10-15 - 51c810e / d565fef1 - lavc 56.6.100 / 56.4.0 - avcodec.h
-  Add AV_HWACCEL_FLAG_IGNORE_LEVEL flag to av_vdpau_bind_context().
-
-2014-10-13 - da21895 / 2df0c32e - lavc 56.5.100 / 56.3.0 - avcodec.h
-  Add AVCodecContext.initial_padding. Deprecate the use of AVCodecContext.delay
-  for audio encoding.
-
-2014-10-08 - bb44f7d / 5a419b2 - lavu 54.10.100 / 54.4.0 - pixdesc.h
-  Add API to return the name of frame and context color properties.
-
-2014-10-06 - a61899a / e3e158e - lavc 56.3.100 / 56.2.0 - vdpau.h
-  Add av_vdpau_bind_context(). This function should now be used for creating
-  (or resetting) a AVVDPAUContext instead of av_vdpau_alloc_context().
-
-2014-10-02 - cdd6f05 - lavc 56.2.100 - avcodec.h
-2014-10-02 - cdd6f05 - lavu 54.9.100 - frame.h
-  Add AV_FRAME_DATA_SKIP_SAMPLES. Add lavc CODEC_FLAG2_SKIP_MANUAL and
-  AVOption "skip_manual", which makes lavc export skip information via
-  AV_FRAME_DATA_SKIP_SAMPLES AVFrame side data, instead of skipping and
-  discarding samples automatically.
-
-2014-10-02 - 0d92b0d - lavu 54.8.100 - avstring.h
-  Add av_match_list()
-
-2014-09-24 - ac68295 - libpostproc 53.1.100
-  Add visualization support
-
-2014-09-19 - 6edd6a4 - lavc 56.1.101 - dv_profile.h
-  deprecate avpriv_dv_frame_profile2(), which was made public by accident.
-
-
--------- 8< --------- FFmpeg 2.4 was cut here -------- 8< ---------
-
-2014-08-25 - 215db29 / b263f8f - lavf 56.3.100 / 56.3.0 - avformat.h
-  Add AVFormatContext.max_ts_probe.
-
-2014-08-28 - f30a815 / 9301486 - lavc 56.1.100 / 56.1.0 - avcodec.h
-  Add AV_PKT_DATA_STEREO3D to export container-level stereo3d information.
-
-2014-08-23 - 8fc9bd0 - lavu 54.7.100 - dict.h
-  AV_DICT_DONT_STRDUP_KEY and AV_DICT_DONT_STRDUP_VAL arguments are now
-  freed even on error. This is consistent with the behaviour all users
-  of it we could find expect.
-
-2014-08-21 - 980a5b0 - lavu 54.6.100 - frame.h motion_vector.h
-  Add AV_FRAME_DATA_MOTION_VECTORS side data and AVMotionVector structure
-
-2014-08-16 - b7d5e01 - lswr 1.1.100 - swresample.h
-  Add AVFrame based API
-
-2014-08-16 - c2829dc - lavu 54.4.100 - dict.h
-  Add av_dict_set_int helper function.
-
-2014-08-13 - c8571c6 / 8ddc326 - lavu 54.3.100 / 54.3.0 - mem.h
-  Add av_strndup().
-
-2014-08-13 - 2ba4577 / a8c104a - lavu 54.2.100 / 54.2.0 - opt.h
-  Add av_opt_get_dict_val/set_dict_val with AV_OPT_TYPE_DICT to support
-  dictionary types being set as options.
-
-2014-08-13 - afbd4b8 - lavf 56.01.0 - avformat.h
-  Add AVFormatContext.event_flags and AVStream.event_flags for signaling to
-  the user when events happen in the file/stream.
-
-2014-08-10 - 78eaaa8 / fb1ddcd - lavr 2.1.0 - avresample.h
-  Add avresample_convert_frame() and avresample_config().
-
-2014-08-10 - 78eaaa8 / fb1ddcd - lavu 54.1.100 / 54.1.0 - error.h
-  Add AVERROR_INPUT_CHANGED and AVERROR_OUTPUT_CHANGED.
-
-2014-08-08 - 3841f2a / d35b94f - lavc 55.73.102 / 55.57.4 - avcodec.h
-  Deprecate FF_IDCT_XVIDMMX define and xvidmmx idct option.
-  Replaced by FF_IDCT_XVID and xvid respectively.
-
-2014-08-08 - 5c3c671 - lavf 55.53.100 - avio.h
-  Add avio_feof() and deprecate url_feof().
-
-2014-08-07 - bb78903 - lsws 2.1.3 - swscale.h
-  sws_getContext is not going to be removed in the future.
-
-2014-08-07 - a561662 / ad1ee5f - lavc 55.73.101 / 55.57.3 - avcodec.h
-  reordered_opaque is not going to be removed in the future.
-
-2014-08-02 - 28a2107 - lavu 52.98.100 - pixelutils.h
-  Add pixelutils API with SAD functions
-
-2014-08-04 - 6017c98 / e9abafc - lavu 52.97.100 / 53.22.0 - pixfmt.h
-  Add AV_PIX_FMT_YA16 pixel format for 16 bit packed gray with alpha.
-
-2014-08-04 - 4c8bc6f / e96c3b8 - lavu 52.96.101 / 53.21.1 - avstring.h
-  Rename AV_PIX_FMT_Y400A to AV_PIX_FMT_YA8 to better identify the format.
-  An alias pixel format and color space name are provided for compatibility.
-
-2014-08-04 - 073c074 / d2962e9 - lavu 52.96.100 / 53.21.0 - pixdesc.h
-  Support name aliases for pixel formats.
-
-2014-08-03 - 71d008e / 1ef9e83 - lavc 55.72.101 / 55.57.2 - avcodec.h
-2014-08-03 - 71d008e / 1ef9e83 - lavu 52.95.100 / 53.20.0 - frame.h
-  Deprecate AVCodecContext.dtg_active_format and use side-data instead.
-
-2014-08-03 - e680c73 - lavc 55.72.100 - avcodec.h
-  Add get_pixels() to AVDCT
-
-2014-08-03 - 9400603 / 9f17685 - lavc 55.71.101 / 55.57.1 - avcodec.h
-  Deprecate unused FF_IDCT_IPP define and ipp avcodec option.
-  Deprecate unused FF_DEBUG_PTS define and pts avcodec option.
-  Deprecate unused FF_CODER_TYPE_DEFLATE define and deflate avcodec option.
-  Deprecate unused FF_DCT_INT define and int avcodec option.
-  Deprecate unused avcodec option scenechange_factor.
-
-2014-07-30 - ba3e331 - lavu 52.94.100 - frame.h
-  Add av_frame_side_data_name()
-
-2014-07-29 - 80a3a66 / 3a19405 - lavf 56.01.100 / 56.01.0 - avformat.h
-  Add mime_type field to AVProbeData, which now MUST be initialized in
-  order to avoid uninitialized reads of the mime_type pointer, likely
-  leading to crashes.
-  Typically, this means you will do 'AVProbeData pd = { 0 };' instead of
-  'AVProbeData pd;'.
-
-2014-07-29 - 31e0b5d / 69e7336 - lavu 52.92.100 / 53.19.0 - avstring.h
-  Make name matching function from lavf public as av_match_name().
-
-2014-07-28 - 2e5c8b0 / c5fca01 - lavc 55.71.100 / 55.57.0 - avcodec.h
-  Add AV_CODEC_PROP_REORDER to mark codecs supporting frame reordering.
-
-2014-07-27 - ff9a154 - lavf 55.50.100 - avformat.h
-  New field int64_t probesize2 instead of deprecated
-  field int probesize.
-
-2014-07-27 - 932ff70 - lavc 55.70.100 - avdct.h
-  Add AVDCT / avcodec_dct_alloc() / avcodec_dct_init().
-
-2014-07-23 - 8a4c086 - lavf 55.49.100 - avio.h
-  Add avio_read_to_bprint()
-
-
--------- 8< --------- FFmpeg 2.3 was cut here -------- 8< ---------
-
-2014-07-14 - 62227a7 - lavf 55.47.100 - avformat.h
-  Add av_stream_get_parser()
-
-2014-07-09 - c67690f / a54f03b - lavu 52.92.100 / 53.18.0 - display.h
-  Add av_display_matrix_flip() to flip the transformation matrix.
-
-2014-07-09 - 1b58f13 / f6ee61f - lavc 55.69.100 / 55.56.0 - dv_profile.h
-  Add a public API for DV profile handling.
-
-2014-06-20 - 0dceefc / 9e500ef - lavu 52.90.100 / 53.17.0 - imgutils.h
-  Add av_image_check_sar().
-
-2014-06-20 - 4a99333 / 874390e - lavc 55.68.100 / 55.55.0 - avcodec.h
-  Add av_packet_rescale_ts() to simplify timestamp conversion.
-
-2014-06-18 - ac293b6 / 194be1f - lavf 55.44.100 / 55.20.0 - avformat.h
-  The proper way for providing a hint about the desired timebase to the muxers
-  is now setting AVStream.time_base, instead of AVStream.codec.time_base as was
-  done previously. The old method is now deprecated.
-
-2014-06-11 - 67d29da - lavc 55.66.101 - avcodec.h
-  Increase FF_INPUT_BUFFER_PADDING_SIZE to 32 due to some corner cases needing
-  it
-
-2014-06-10 - 5482780 - lavf 55.43.100 - avformat.h
-  New field int64_t max_analyze_duration2 instead of deprecated
-  int max_analyze_duration.
-
-2014-05-30 - 00759d7 - lavu 52.89.100 - opt.h
-  Add av_opt_copy()
-
-2014-06-01 - 03bb99a / 0957b27 - lavc 55.66.100 / 55.54.0 - avcodec.h
-  Add AVCodecContext.side_data_only_packets to allow encoders to output packets
-  with only side data. This option may become mandatory in the future, so all
-  users are recommended to update their code and enable this option.
-
-2014-06-01 - 6e8e9f1 / 8c02adc - lavu 52.88.100 / 53.16.0 - frame.h, pixfmt.h
-  Move all color-related enums (AVColorPrimaries, AVColorSpace, AVColorRange,
-  AVColorTransferCharacteristic, and AVChromaLocation) inside lavu.
-  And add AVFrame fields for them.
-
-2014-05-29 - bdb2e80 / b2d4565 - lavr 1.3.0 - avresample.h
-  Add avresample_max_output_samples
-
-2014-05-28 - d858ee7 / 6d21259 - lavf 55.42.100 / 55.19.0 - avformat.h
-  Add strict_std_compliance and related AVOptions to support experimental
-  muxing.
-
-2014-05-26 - 55cc60c - lavu 52.87.100 - threadmessage.h
-  Add thread message queue API.
-
-2014-05-26 - c37d179 - lavf 55.41.100 - avformat.h
-  Add format_probesize to AVFormatContext.
-
-2014-05-20 - 7d25af1 / c23c96b - lavf 55.39.100 / 55.18.0 - avformat.h
-  Add av_stream_get_side_data() to access stream-level side data
-  in the same way as av_packet_get_side_data().
-
-2014-05-20 - 7336e39 - lavu 52.86.100 - fifo.h
-  Add av_fifo_alloc_array() function.
-
-2014-05-19 - ef1d4ee / bddd8cb - lavu 52.85.100 / 53.15.0 - frame.h, display.h
-  Add AV_FRAME_DATA_DISPLAYMATRIX for exporting frame-level
-  spatial rendering on video frames for proper display.
-
-2014-05-19 - ef1d4ee / bddd8cb - lavc 55.64.100 / 55.53.0 - avcodec.h
-  Add AV_PKT_DATA_DISPLAYMATRIX for exporting packet-level
-  spatial rendering on video frames for proper display.
-
-2014-05-19 - 999a99c / a312f71 - lavf 55.38.101 / 55.17.1 - avformat.h
-  Deprecate AVStream.pts and the AVFrac struct, which was its only use case.
-  See use av_stream_get_end_pts()
-
-2014-05-18 - 68c0518 / fd05602 - lavc 55.63.100 / 55.52.0 - avcodec.h
-  Add avcodec_free_context(). From now on it should be used for freeing
-  AVCodecContext.
-
-2014-05-17 - 0eec06e / 1bd0bdc - lavu 52.84.100 / 54.5.0 - time.h
-  Add av_gettime_relative() av_gettime_relative_is_monotonic()
-
-2014-05-15 - eacf7d6 / 0c1959b - lavf 55.38.100 / 55.17.0 - avformat.h
-  Add AVFMT_FLAG_BITEXACT flag. Muxers now use it instead of checking
-  CODEC_FLAG_BITEXACT on the first stream.
-
-2014-05-15 - 96cb4c8 - lswr 0.19.100 - swresample.h
-  Add swr_close()
-
-2014-05-11 - 14aef38 / 66e6c8a - lavu 52.83.100 / 53.14.0 - pixfmt.h
-  Add AV_PIX_FMT_VDA for new-style VDA acceleration.
-
-2014-05-07 - 351f611 - lavu 52.82.100 - fifo.h
-  Add av_fifo_freep() function.
-
-2014-05-02 - ba52fb11 - lavu 52.81.100 - opt.h
-  Add av_opt_set_dict2() function.
-
-2014-05-01 - e77b985 / a2941c8 - lavc 55.60.103 / 55.50.3 - avcodec.h
-  Deprecate CODEC_FLAG_MV0. It is replaced by the flag "mv0" in the
-  "mpv_flags" private option of the mpegvideo encoders.
-
-2014-05-01 - e40ae8c / 6484149 - lavc 55.60.102 / 55.50.2 - avcodec.h
-  Deprecate CODEC_FLAG_GMC. It is replaced by the "gmc" private option of the
-  libxvid encoder.
-
-2014-05-01 - 1851643 / b2c3171 - lavc 55.60.101 / 55.50.1 - avcodec.h
-  Deprecate CODEC_FLAG_NORMALIZE_AQP. It is replaced by the flag "naq" in the
-  "mpv_flags" private option of the mpegvideo encoders.
-
-2014-05-01 - cac07d0 / 5fcceda - avcodec.h
-  Deprecate CODEC_FLAG_INPUT_PRESERVED. Its functionality is replaced by passing
-  reference-counted frames to encoders.
-
-2014-04-30 - 617e866 - lavu 52.81.100 - pixdesc.h
-  Add av_find_best_pix_fmt_of_2(), av_get_pix_fmt_loss()
-  Deprecate avcodec_get_pix_fmt_loss(), avcodec_find_best_pix_fmt_of_2()
-
-2014-04-29 - 1bf6396 - lavc 55.60.100 - avcodec.h
-  Add AVCodecDescriptor.mime_types field.
-
-2014-04-29 - b804eb4 - lavu 52.80.100 - hash.h
-  Add av_hash_final_bin(), av_hash_final_hex() and av_hash_final_b64().
-
-2014-03-07 - 8b2a130 - lavc 55.50.0 / 55.53.100 - dxva2.h
-  Add FF_DXVA2_WORKAROUND_INTEL_CLEARVIDEO for old Intel GPUs.
-
-2014-04-22 - 502512e /dac7e8a - lavu 53.13.0 / 52.78.100 - avutil.h
-  Add av_get_time_base_q().
-
-2014-04-17 - a8d01a7 / 0983d48 - lavu 53.12.0 / 52.77.100 - crc.h
-  Add AV_CRC_16_ANSI_LE crc variant.
-
-2014-04-15 - ef818d8 - lavf 55.37.101 - avformat.h
-  Add av_format_inject_global_side_data()
-
-2014-04-12 - 4f698be - lavu 52.76.100 - log.h
-  Add av_log_get_flags()
-
-2014-04-11 - 6db42a2b - lavd 55.12.100 - avdevice.h
-  Add avdevice_capabilities_create() function.
-  Add avdevice_capabilities_free() function.
-
-2014-04-07 - 0a1cc04 / 8b17243 - lavu 52.75.100 / 53.11.0 - pixfmt.h
-  Add AV_PIX_FMT_YVYU422 pixel format.
-
-2014-04-04 - c1d0536 / 8542f9c - lavu 52.74.100 / 53.10.0 - replaygain.h
-  Full scale for peak values is now 100000 (instead of UINT32_MAX) and values
-  may overflow.
-
-2014-04-03 - c16e006 / 7763118 - lavu 52.73.100 / 53.9.0 - log.h
-  Add AV_LOG(c) macro to have 256 color debug messages.
-
-2014-04-03 - eaed4da9 - lavu 52.72.100 - opt.h
-  Add AV_OPT_MULTI_COMPONENT_RANGE define to allow return
-  multi-component option ranges.
-
-2014-03-29 - cd50a44b - lavu 52.70.100 - mem.h
-  Add av_dynarray_add_nofree() function.
-
-2014-02-24 - 3e1f241 / d161ae0 - lavu 52.69.100 / 53.8.0 - frame.h
-  Add av_frame_remove_side_data() for removing a single side data
-  instance from a frame.
-
-2014-03-24 - 83e8978 / 5a7e35d - lavu 52.68.100 / 53.7.0 - frame.h, replaygain.h
-  Add AV_FRAME_DATA_REPLAYGAIN for exporting replaygain tags.
-  Add a new header replaygain.h with the AVReplayGain struct.
-
-2014-03-24 - 83e8978 / 5a7e35d - lavc 55.54.100 / 55.36.0 - avcodec.h
-  Add AV_PKT_DATA_REPLAYGAIN for exporting replaygain tags.
-
-2014-03-24 - 595ba3b / 25b3258 - lavf 55.35.100 / 55.13.0 - avformat.h
-  Add AVStream.side_data and AVStream.nb_side_data for exporting stream-global
-  side data (e.g. replaygain tags, video rotation)
-
-2014-03-24 - bd34e26 / 0e2c3ee - lavc 55.53.100 / 55.35.0 - avcodec.h
-  Give the name AVPacketSideData to the previously anonymous struct used for
-  AVPacket.side_data.
-
-
--------- 8< --------- FFmpeg 2.2 was cut here -------- 8< ---------
-
-2014-03-18 - 37c07d4 - lsws 2.5.102
-  Make gray16 full-scale.
-
-2014-03-16 - 6b1ca17 / 1481d24 - lavu 52.67.100 / 53.6.0 - pixfmt.h
-  Add RGBA64_LIBAV pixel format and variants for compatibility
-
-2014-03-11 - 3f3229c - lavf 55.34.101 - avformat.h
-  Set AVFormatContext.start_time_realtime when demuxing.
-
-2014-03-03 - 06fed440 - lavd 55.11.100 - avdevice.h
-  Add av_input_audio_device_next().
-  Add av_input_video_device_next().
-  Add av_output_audio_device_next().
-  Add av_output_video_device_next().
-
-2014-02-24 - fff5262 / 1155fd0 - lavu 52.66.100 / 53.5.0 - frame.h
-  Add av_frame_copy() for copying the frame data.
-
-2014-02-24 - a66be60 - lswr 0.18.100 - swresample.h
-  Add swr_is_initialized() for checking whether a resample context is initialized.
-
-2014-02-22 - 5367c0b / 7e86c27 - lavr 1.2.0 - avresample.h
-  Add avresample_is_open() for checking whether a resample context is open.
-
-2014-02-19 - 6a24d77 / c3ecd96 - lavu 52.65.100 / 53.4.0  - opt.h
-  Add AV_OPT_FLAG_EXPORT and AV_OPT_FLAG_READONLY to mark options meant (only)
-  for reading.
-
-2014-02-19 - f4c8d00 / 6bb8720 - lavu 52.64.101 / 53.3.1 - opt.h
-  Deprecate unused AV_OPT_FLAG_METADATA.
-
-2014-02-16 - 81c3f81 - lavd 55.10.100 - avdevice.h
-  Add avdevice_list_devices() and avdevice_free_list_devices()
-
-2014-02-16 - db3c970 - lavf 55.33.100 - avio.h
-  Add avio_find_protocol_name() to find out the name of the protocol that would
-  be selected for a given URL.
-
-2014-02-15 - a2bc6c1 / c98f316 - lavu 52.64.100 / 53.3.0 - frame.h
-  Add AV_FRAME_DATA_DOWNMIX_INFO value to the AVFrameSideDataType enum and
-  downmix_info.h API, which identify downmix-related metadata.
-
-2014-02-11 - 1b05ac2 - lavf 55.32.100 - avformat.h
-  Add av_write_uncoded_frame() and av_interleaved_write_uncoded_frame().
-
-2014-02-04 - 3adb5f8 / d9ae103 - lavf 55.30.100 / 55.11.0 - avformat.h
-  Add AVFormatContext.max_interleave_delta for controlling amount of buffering
-  when interleaving.
-
-2014-02-02 - 5871ee5 - lavf 55.29.100 - avformat.h
-  Add output_ts_offset muxing option to AVFormatContext.
-
-2014-01-27 - 102bd64 - lavd 55.7.100 - avdevice.h
-                       lavf 55.28.100 - avformat.h
-  Add avdevice_dev_to_app_control_message() function.
-
-2014-01-27 - 7151411 - lavd 55.6.100 - avdevice.h
-                       lavf 55.27.100 - avformat.h
-  Add avdevice_app_to_dev_control_message() function.
-
-2014-01-24 - 86bee79 - lavf 55.26.100 - avformat.h
-  Add AVFormatContext option metadata_header_padding to allow control over the
-  amount of padding added.
-
-2014-01-20 - eef74b2 / 93c553c - lavc 55.48.102 / 55.32.1 - avcodec.h
-  Edges are not required anymore on video buffers allocated by get_buffer2()
-  (i.e. as if the CODEC_FLAG_EMU_EDGE flag was always on). Deprecate
-  CODEC_FLAG_EMU_EDGE and avcodec_get_edge_width().
-
-2014-01-19 - 1a193c4 - lavf 55.25.100 - avformat.h
-  Add avformat_get_mov_video_tags() and avformat_get_mov_audio_tags().
-
-2014-01-19 - 3532dd5 - lavu 52.63.100 - rational.h
-  Add av_make_q() function.
-
-2014-01-05 - 4cf4da9 / 5b4797a - lavu 52.62.100 / 53.2.0 - frame.h
-  Add AV_FRAME_DATA_MATRIXENCODING value to the AVFrameSideDataType enum, which
-  identifies AVMatrixEncoding data.
-
-2014-01-05 - 751385f / 5c437fb - lavu 52.61.100 / 53.1.0 - channel_layout.h
-  Add values for various Dolby flags to the AVMatrixEncoding enum.
-
-2014-01-04 - b317f94 - lavu 52.60.100 - mathematics.h
-  Add av_add_stable() function.
-
-2013-12-22 - 911676c - lavu 52.59.100 - avstring.h
-  Add av_strnlen() function.
-
-2013-12-09 - 64f73ac - lavu 52.57.100 - opencl.h
-  Add av_opencl_benchmark() function.
-
-2013-11-30 - 82b2e9c - lavu 52.56.100 - ffversion.h
-  Moves version.h to libavutil/ffversion.h.
-  Install ffversion.h and make it public.
-
-2013-12-11 - 29c83d2 / b9fb59d,409a143 / 9431356,44967ab / d7b3ee9 - lavc 55.45.101 / 55.28.1 - avcodec.h
-  av_frame_alloc(), av_frame_unref() and av_frame_free() now can and should be
-  used instead of avcodec_alloc_frame(), avcodec_get_frame_defaults() and
-  avcodec_free_frame() respectively. The latter three functions are deprecated.
-
-2013-12-09 - 7a60348 / 7e244c6- - lavu 52.58.100 / 52.20.0 - frame.h
-  Add AV_FRAME_DATA_STEREO3D value to the AVFrameSideDataType enum and
-  stereo3d.h API, that identify codec-independent stereo3d information.
-
-2013-11-26 - 625b290 / 1eaac1d- - lavu 52.55.100 / 52.19.0 - frame.h
-  Add AV_FRAME_DATA_A53_CC value to the AVFrameSideDataType enum, which
-  identifies ATSC A53 Part 4 Closed Captions data.
-
-2013-11-22 - 6859065 - lavu 52.54.100 - avstring.h
-  Add av_utf8_decode() function.
-
-2013-11-22 - fb7d70c - lavc 55.44.100 - avcodec.h
-  Add HEVC profiles
-
-2013-11-20 - c28b61c - lavc 55.44.100 - avcodec.h
-  Add av_packet_{un,}pack_dictionary()
-  Add AV_PKT_METADATA_UPDATE side data type, used to transmit key/value
-  strings between a stream and the application.
-
-2013-11-14 - 7c888ae / cce3e0a - lavu 52.53.100 / 52.18.0 - mem.h
-  Move av_fast_malloc() and av_fast_realloc() for libavcodec to libavutil.
-
-2013-11-14 - b71e4d8 / 8941971 - lavc 55.43.100 / 55.27.0 - avcodec.h
-  Deprecate AVCodecContext.error_rate, it is replaced by the 'error_rate'
-  private option of the mpegvideo encoder family.
-
-2013-11-14 - 31c09b7 / 728c465 - lavc 55.42.100 / 55.26.0 - vdpau.h
-  Add av_vdpau_get_profile().
-  Add av_vdpau_alloc_context(). This function must from now on be
-  used for allocating AVVDPAUContext.
-
-2013-11-04 - be41f21 / cd8f772 - lavc 55.41.100 / 55.25.0 - avcodec.h
-                       lavu 52.51.100 - frame.h
-  Add ITU-R BT.2020 and other not yet included values to color primaries,
-  transfer characteristics and colorspaces.
-
-2013-11-04 - 85cabf1 - lavu 52.50.100 - avutil.h
-  Add av_fopen_utf8()
-
-2013-10-31 - 78265fc / 28096e0 - lavu 52.49.100 / 52.17.0 - frame.h
-  Add AVFrame.flags and AV_FRAME_FLAG_CORRUPT.
-
-
--------- 8< --------- FFmpeg 2.1 was cut here -------- 8< ---------
-
-2013-10-27 - dbe6f9f - lavc 55.39.100 - avcodec.h
+2013-10-27 - xxxxxxx - lavc 55.39.100 - avcodec.h
   Add CODEC_CAP_DELAY support to avcodec_decode_subtitle2.
 
-2013-10-27 - d61617a - lavu 52.48.100 - parseutils.h
+2013-10-27 - xxxxxxx - lavu 52.48.100 - parseutils.h
   Add av_get_known_color_name().
 
-2013-10-17 - 8696e51 - lavu 52.47.100 - opt.h
+2013-10-17 - xxxxxxx - lavu 52.47.100 - opt.h
   Add AV_OPT_TYPE_CHANNEL_LAYOUT and channel layout option handlers
   av_opt_get_channel_layout() and av_opt_set_channel_layout().
 
-2013-10-06 - ccf96f8 -libswscale 2.5.101 - options.c
+2013-10-xx - xxxxxxx -libswscale 2.5.101 - options.c
   Change default scaler to bicubic
 
-2013-10-03 - e57dba0 - lavc 55.34.100 - avcodec.h
+2013-10-03 - xxxxxxx - lavc 55.34.100 - avcodec.h
   Add av_codec_get_max_lowres()
 
-2013-10-02 - 5082fcc - lavf 55.19.100 - avformat.h
+2013-10-02 - xxxxxxx - lavf 55.19.100 - avformat.h
   Add audio/video/subtitle AVCodec fields to AVFormatContext to force specific
   decoders
 
-2013-09-28 - 7381d31 / 0767bfd - lavfi 3.88.100 / 3.11.0 - avfilter.h
+2013-08-xx - xxxxxxx - lavfi 3.11.0 - avfilter.h
   Add AVFilterGraph.execute and AVFilterGraph.opaque for custom slice threading
   implementations.
 
-2013-09-21 - 85f8a3c / e208e6d - lavu 52.46.100 / 52.16.0 - pixfmt.h
+2013-09-21 - xxxxxxx - lavu 52.16.0 - pixfmt.h
   Add interleaved 4:2:2 8/10-bit formats AV_PIX_FMT_NV16 and
   AV_PIX_FMT_NV20.
 
@@ -578,7 +49,7 @@ API changes, most recent first:
 2013-09-04 - 3e1f507 - lavc 55.31.101 - avcodec.h
   avcodec_close() argument can be NULL.
 
-2013-09-04 - 36cd017a - lavf 55.16.101 - avformat.h
+2013-09-04 - 36cd017 - lavf 55.16.101 - avformat.h
   avformat_close_input() argument can be NULL and point on NULL.
 
 2013-08-29 - e31db62 - lavf 55.15.100 - avformat.h
@@ -587,10 +58,10 @@ API changes, most recent first:
 2013-08-15 - 1e0e193 - lsws 2.5.100 -
   Add a sws_dither AVOption, allowing to set the dither algorithm used
 
-2013-08-11 - d404fe35 - lavc 55.27.100 - vdpau.h
+2013-08-xx - xxxxxxx - lavc 55.27.100 - vdpau.h
   Add a render2 alternative to the render callback function.
 
-2013-08-11 - af05edc - lavc 55.26.100 - vdpau.h
+2013-08-xx - xxxxxxx - lavc 55.26.100 - vdpau.h
   Add allocation function for AVVDPAUContext, allowing
   to extend it in the future without breaking ABI/API.
 
@@ -600,7 +71,7 @@ API changes, most recent first:
 
 2013-08-05 - 9547e3e / f824535 - lavc 55.22.100 / 55.13.0 - avcodec.h
   Deprecate the bitstream-related members from struct AVVDPAUContext.
-  The bitstream buffers no longer need to be explicitly freed.
+  The bistream buffers no longer need to be explicitly freed.
 
 2013-08-05 - 3b805dc / 549294f - lavc 55.21.100 / 55.12.0 - avcodec.h
   Deprecate the CODEC_CAP_HWACCEL_VDPAU codec capability. Use CODEC_CAP_HWACCEL
@@ -616,9 +87,6 @@ API changes, most recent first:
   Add avcodec_chroma_pos_to_enum()
   Add avcodec_enum_to_chroma_pos()
 
-
--------- 8< --------- FFmpeg 2.0 was cut here -------- 8< ---------
-
 2013-07-03 - 838bd73 - lavfi 3.78.100 - avfilter.h
   Deprecate avfilter_graph_parse() in favor of the equivalent
   avfilter_graph_parse_ptr().
@@ -691,9 +159,6 @@ API changes, most recent first:
 2013-03-17 - 7aa9af5 - lavu 52.20.100 - opt.h
   Add AV_OPT_TYPE_VIDEO_RATE value to AVOptionType enum.
 
-
--------- 8< --------- FFmpeg 1.2 was cut here -------- 8< ---------
-
 2013-03-07 - 9767ec6 - lavu 52.18.100 - avstring.h,bprint.h
   Add av_escape() and av_bprint_escape() API.
 
@@ -706,9 +171,6 @@ API changes, most recent first:
 2013-01-01 - 2eb2e17 - lavfi 3.34.100
   Add avfilter_get_audio_buffer_ref_from_arrays_channels.
 
-
--------- 8< --------- FFmpeg 1.1 was cut here -------- 8< ---------
-
 2012-12-20 - 34de47aa - lavfi 3.29.100 - avfilter.h
   Add AVFilterLink.channels, avfilter_link_get_channels()
   and avfilter_ref_get_channels().
@@ -754,9 +216,6 @@ API changes, most recent first:
   Add LIBSWRESAMPLE_VERSION, LIBSWRESAMPLE_BUILD
   and LIBSWRESAMPLE_IDENT symbols.
 
-
--------- 8< --------- FFmpeg 1.0 was cut here -------- 8< ---------
-
 2012-09-06 - 29e972f - lavu 51.72.100 - parseutils.h
   Add av_small_strptime() time parsing function.
 
@@ -961,9 +420,6 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
   avresample_read() are now uint8_t** instead of void**.
   Libavresample is now stable.
 
-2012-09-26 - 3ba0dab7 / 1384df64 - lavf 54.29.101 / 56.06.3 - avformat.h
-  Add AVFormatContext.avoid_negative_ts.
-
 2012-09-24 - 46a3595 / a42aada - lavc 54.59.100 / 54.28.0 - avcodec.h
   Add avcodec_free_frame(). This function must now
   be used for freeing an AVFrame.
@@ -1178,9 +634,6 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
 2012-01-12 - b18e17e / 3167dc9 - lavfi 2.59.100 / 2.15.0
   Add a new installed header -- libavfilter/version.h -- with version macros.
 
-
--------- 8< --------- FFmpeg 0.9 was cut here -------- 8< ---------
-
 2011-12-08 - a502939 - lavfi 2.52.0
   Add av_buffersink_poll_frame() to buffersink.h.
 
@@ -1209,9 +662,6 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
   Add avformat_close_input().
   Deprecate av_close_input_file() and av_close_input_stream().
 
-2011-12-09 - c59b80c / b2890f5 - lavu 51.32.0 / 51.20.0 - audioconvert.h
-  Expand the channel layout list.
-
 2011-12-02 - e4de716 / 0eea212 - lavc 53.40.0 / 53.25.0
   Add nb_samples and extended_data fields to AVFrame.
   Deprecate AVCODEC_MAX_AUDIO_FRAME_SIZE.
@@ -1225,10 +675,6 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
   Change AVCodecContext.error[4] to [8] at next major bump.
   Add AV_NUM_DATA_POINTERS to simplify the bump transition.
 
-2011-11-24 - lavu 51.29.0 / 51.19.0
-  92afb43 / bd97b2e - add planar RGB pixel formats
-  92afb43 / 6b0768e - add PIX_FMT_PLANAR and PIX_FMT_RGB pixel descriptions
-
 2011-11-23 - 8e576d5 / bbb46f3 - lavu 51.27.0 / 51.18.0
   Add av_samples_get_buffer_size(), av_samples_fill_arrays(), and
   av_samples_alloc(), to samplefmt.h.
@@ -1390,13 +836,6 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
 2011-06-28 - 5129336 - lavu 51.11.0 - avutil.h
   Define the AV_PICTURE_TYPE_NONE value in AVPictureType enum.
 
-
--------- 8< --------- FFmpeg 0.7 was cut here -------- 8< ---------
-
-
-
--------- 8< --------- FFmpeg 0.8 was cut here -------- 8< ---------
-
 2011-06-19 - fd2c0a5 - lavfi 2.23.0 - avfilter.h
   Add layout negotiation fields and helper functions.
 
@@ -2074,9 +1513,6 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
 2010-06-02 - 7e566bb - lavc 52.73.0 - av_get_codec_tag_string()
   Add av_get_codec_tag_string().
 
-
--------- 8< --------- FFmpeg 0.6 was cut here -------- 8< ---------
-
 2010-06-01 - 2b99142 - lsws 0.11.0 - convertPalette API
   Add sws_convertPalette8ToPacked32() and sws_convertPalette8ToPacked24().
 
@@ -2094,6 +1530,10 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
 2010-05-09 - b6bc205 - lavfi 1.20.0 - AVFilterPicRef
   Add interlaced and top_field_first fields to AVFilterPicRef.
 
+------------------------------8<-------------------------------------
+                   0.6 branch was cut here
+----------------------------->8--------------------------------------
+
 2010-05-01 - 8e2ee18 - lavf 52.62.0 - probe function
   Add av_probe_input_format2 to API, it allows ignoring probe
   results below given score and returns the actual probe score.

doc/Doxyfile

@@ -31,7 +31,7 @@ PROJECT_NAME           = FFmpeg
 # This could be handy for archiving the generated documentation or
 # if some version control system is used.
 
-PROJECT_NUMBER         = 2.6.5
+PROJECT_NUMBER         = 2.1.4
 
 # With the PROJECT_LOGO tag one can specify a logo or icon that is included
 # in the documentation. The maximum height of the logo should not exceed 55
@@ -759,7 +759,7 @@ ALPHABETICAL_INDEX     = YES
 # the COLS_IN_ALPHA_INDEX tag can be used to specify the number of columns
 # in which this list will be split (can be a number in the range [1..20])
 
-COLS_IN_ALPHA_INDEX    = 5
+COLS_IN_ALPHA_INDEX    = 2
 
 # In case all classes in a project start with a common prefix, all
 # classes will be put under the same header in the alphabetical index.
@@ -793,13 +793,13 @@ HTML_FILE_EXTENSION    = .html
 # each generated HTML page. If it is left blank doxygen will generate a
 # standard header.
 
-HTML_HEADER            =
+#HTML_HEADER            = doc/doxy/header.html
 
 # The HTML_FOOTER tag can be used to specify a personal HTML footer for
 # each generated HTML page. If it is left blank doxygen will generate a
 # standard footer.
 
-HTML_FOOTER            =
+#HTML_FOOTER            = doc/doxy/footer.html
 
 # The HTML_STYLESHEET tag can be used to specify a user-defined cascading
 # style sheet that is used by each HTML page. It can be used to
@@ -808,7 +808,7 @@ HTML_FOOTER            =
 # the style sheet file to the HTML output directory, so don't put your own
 # stylesheet in the HTML output directory as well, or it will be erased!
 
-HTML_STYLESHEET        =
+#HTML_STYLESHEET        = doc/doxy/doxy_stylesheet.css
 
 # The HTML_COLORSTYLE_HUE tag controls the color of the HTML output.
 # Doxygen will adjust the colors in the stylesheet and background images
@@ -1056,7 +1056,7 @@ FORMULA_TRANSPARENT    = YES
 # typically be disabled. For large projects the javascript based search engine
 # can be slow, then enabling SERVER_BASED_SEARCH may provide a better solution.
 
-SEARCHENGINE           = YES
+SEARCHENGINE           = NO
 
 # When the SERVER_BASED_SEARCH tag is enabled the search engine will be
 # implemented using a PHP enabled web server instead of at the web client
@@ -1359,8 +1359,6 @@ PREDEFINED             = "__attribute__(x)=" \
                          "DECLARE_ALIGNED(a,t,n)=t n" \
                          "offsetof(x,y)=0x42" \
                          av_alloc_size \
-                         AV_GCC_VERSION_AT_LEAST(x,y)=1 \
-                         __GNUC__=1 \
 
 # If the MACRO_EXPANSION and EXPAND_ONLY_PREDEF tags are set to YES then
 # this tag can be used to specify a list of macro names that should be expanded.

doc/Makefile

@@ -14,11 +14,11 @@ COMPONENTS-$(CONFIG_AVFORMAT)   += ffmpeg-formats ffmpeg-protocols
 COMPONENTS-$(CONFIG_AVDEVICE)   += ffmpeg-devices
 COMPONENTS-$(CONFIG_AVFILTER)   += ffmpeg-filters
 
-MANPAGES1   = $(AVPROGS-yes:%=doc/%.1)    $(AVPROGS-yes:%=doc/%-all.1)    $(COMPONENTS-yes:%=doc/%.1)
+MANPAGES1   = $(PROGS-yes:%=doc/%.1)    $(PROGS-yes:%=doc/%-all.1)    $(COMPONENTS-yes:%=doc/%.1)
 MANPAGES3   = $(LIBRARIES-yes:%=doc/%.3)
 MANPAGES    = $(MANPAGES1) $(MANPAGES3)
-PODPAGES    = $(AVPROGS-yes:%=doc/%.pod)  $(AVPROGS-yes:%=doc/%-all.pod)  $(COMPONENTS-yes:%=doc/%.pod)  $(LIBRARIES-yes:%=doc/%.pod)
-HTMLPAGES   = $(AVPROGS-yes:%=doc/%.html) $(AVPROGS-yes:%=doc/%-all.html) $(COMPONENTS-yes:%=doc/%.html) $(LIBRARIES-yes:%=doc/%.html) \
+PODPAGES    = $(PROGS-yes:%=doc/%.pod)  $(PROGS-yes:%=doc/%-all.pod)  $(COMPONENTS-yes:%=doc/%.pod)  $(LIBRARIES-yes:%=doc/%.pod)
+HTMLPAGES   = $(PROGS-yes:%=doc/%.html) $(PROGS-yes:%=doc/%-all.html) $(COMPONENTS-yes:%=doc/%.html) $(LIBRARIES-yes:%=doc/%.html) \
               doc/developer.html                                        \
               doc/faq.html                                              \
               doc/fate.html                                             \
@@ -36,29 +36,6 @@ DOCS-$(CONFIG_MANPAGES)  += $(MANPAGES)
 DOCS-$(CONFIG_TXTPAGES)  += $(TXTPAGES)
 DOCS = $(DOCS-yes)
 
-DOC_EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE)      += avio_reading
-DOC_EXAMPLES-$(CONFIG_AVCODEC_EXAMPLE)           += avcodec
-DOC_EXAMPLES-$(CONFIG_DECODING_ENCODING_EXAMPLE) += decoding_encoding
-DOC_EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding
-DOC_EXAMPLES-$(CONFIG_EXTRACT_MVS_EXAMPLE)       += extract_mvs
-DOC_EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE)      += filter_audio
-DOC_EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE)   += filtering_audio
-DOC_EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE)   += filtering_video
-DOC_EXAMPLES-$(CONFIG_METADATA_EXAMPLE)          += metadata
-DOC_EXAMPLES-$(CONFIG_MUXING_EXAMPLE)            += muxing
-DOC_EXAMPLES-$(CONFIG_QSVDEC_EXAMPLE)            += qsvdec
-DOC_EXAMPLES-$(CONFIG_REMUXING_EXAMPLE)          += remuxing
-DOC_EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE)  += resampling_audio
-DOC_EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE)     += scaling_video
-DOC_EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE)     += transcode_aac
-DOC_EXAMPLES-$(CONFIG_TRANSCODING_EXAMPLE)       += transcoding
-ALL_DOC_EXAMPLES_LIST = $(DOC_EXAMPLES-) $(DOC_EXAMPLES-yes)
-
-DOC_EXAMPLES       := $(DOC_EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)$(EXESUF))
-ALL_DOC_EXAMPLES   := $(ALL_DOC_EXAMPLES_LIST:%=doc/examples/%$(PROGSSUF)$(EXESUF))
-ALL_DOC_EXAMPLES_G := $(ALL_DOC_EXAMPLES_LIST:%=doc/examples/%$(PROGSSUF)_g$(EXESUF))
-PROGS              += $(DOC_EXAMPLES)
-
 all-$(CONFIG_DOC): doc
 
 doc: documentation
@@ -66,9 +43,7 @@ doc: documentation
 apidoc: doc/doxy/html
 documentation: $(DOCS)
 
-examples: $(DOC_EXAMPLES)
-
-TEXIDEP = perl $(SRC_PATH)/doc/texidep.pl $(SRC_PATH) $< $@ >$(@:%=%.d)
+TEXIDEP = awk '/^@(verbatim)?include/ { printf "$@: $(@D)/%s\n", $$2 }' <$< >$(@:%=%.d)
 
 doc/%.txt: TAG = TXT
 doc/%.txt: doc/%.texi
@@ -83,25 +58,14 @@ $(GENTEXI): doc/avoptions_%.texi: doc/print_options$(HOSTEXESUF)
 	$(M)doc/print_options $* > $@
 
 doc/%.html: TAG = HTML
-doc/%-all.html: TAG = HTML
-
-ifdef HAVE_MAKEINFO_HTML
-doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.pm $(GENTEXI)
-	$(Q)$(TEXIDEP)
-	$(M)makeinfo --html -I doc --no-split -D config-not-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $<
-
-doc/%-all.html: doc/%.texi $(SRC_PATH)/doc/t2h.pm $(GENTEXI)
-	$(Q)$(TEXIDEP)
-	$(M)makeinfo --html -I doc --no-split -D config-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $<
-else
 doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
 	$(Q)$(TEXIDEP)
 	$(M)texi2html -I doc -monolithic --D=config-not-all --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
 
+doc/%-all.html: TAG = HTML
 doc/%-all.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
 	$(Q)$(TEXIDEP)
 	$(M)texi2html -I doc -monolithic --D=config-all --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
-endif
 
 doc/%.pod: TAG = POD
 doc/%.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)
@@ -115,19 +79,14 @@ doc/%-all.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)
 
 doc/%.1 doc/%.3: TAG = MAN
 doc/%.1: doc/%.pod $(GENTEXI)
-	$(M)pod2man --section=1 --center=" " --release=" " --date=" " $< > $@
+	$(M)pod2man --section=1 --center=" " --release=" " $< > $@
 doc/%.3: doc/%.pod $(GENTEXI)
-	$(M)pod2man --section=3 --center=" " --release=" " --date=" " $< > $@
+	$(M)pod2man --section=3 --center=" " --release=" " $< > $@
 
 $(DOCS) doc/doxy/html: | doc/
-$(DOC_EXAMPLES:%$(EXESUF)=%.o): | doc/examples
-OBJDIRS += doc/examples
 
-DOXY_INPUT      = $(addprefix $(SRC_PATH)/, $(INSTHEADERS) $(DOC_EXAMPLES:%$(EXESUF)=%.c) $(LIB_EXAMPLES:%$(EXESUF)=%.c))
-
-doc/doxy/html: TAG = DOXY
-doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(SRC_PATH)/doc/doxy-wrapper.sh $(DOXY_INPUT)
-	$(M)$(SRC_PATH)/doc/doxy-wrapper.sh $(SRC_PATH) $< $(DOXYGEN) $(DOXY_INPUT)
+doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(INSTHEADERS)
+	$(M)$(SRC_PATH)/doc/doxy-wrapper.sh $(SRC_PATH) $^
 
 install-doc: install-html install-man
 
@@ -161,7 +120,7 @@ uninstall-html:
 	$(RM) -r "$(DOCDIR)"
 
 uninstall-man:
-	$(RM) $(addprefix "$(MANDIR)/man1/",$(AVPROGS-yes:%=%.1) $(AVPROGS-yes:%=%-all.1) $(COMPONENTS-yes:%=%.1))
+	$(RM) $(addprefix "$(MANDIR)/man1/",$(PROGS-yes:%=%.1) $(PROGS-yes:%=%-all.1) $(COMPONENTS-yes:%=%.1))
 	$(RM) $(addprefix "$(MANDIR)/man3/",$(LIBRARIES-yes:%=%.3))
 
 clean:: docclean
@@ -169,13 +128,8 @@ clean:: docclean
 distclean:: docclean
 	$(RM) doc/config.texi
 
-examplesclean:
-	$(RM) $(ALL_DOC_EXAMPLES) $(ALL_DOC_EXAMPLES_G)
-	$(RM) $(CLEANSUFFIXES:%=doc/examples/%)
-
-docclean: examplesclean
-	$(RM) $(CLEANSUFFIXES:%=doc/%)
-	$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 doc/*.3 doc/avoptions_*.texi
+docclean:
+	$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 doc/*.3 $(CLEANSUFFIXES:%=doc/%) doc/avoptions_*.texi
 	$(RM) -r doc/doxy/html
 
 -include $(wildcard $(DOCS:%=%.d))

doc/RELEASE_NOTES

@@ -0,0 +1,16 @@
+Release Notes
+=============
+
+* 2.1 "Fourier"  October, 2013
+
+
+General notes
+-------------
+See the Changelog file for a list of significant changes. Note, there
+are many more new features and bugfixes than whats listed there.
+
+Bugreports against FFmpeg git master or the most recent FFmpeg release are
+accepted. If you are experiencing issues with any formally released version of
+FFmpeg, please try git master to check if the issue still exists. If it does,
+make your report against the development code following the usual bug reporting
+guidelines.

doc/avutil.txt

@@ -0,0 +1,36 @@
+AVUtil
+======
+libavutil is a small lightweight library of generally useful functions.
+It is not a library for code needed by both libavcodec and libavformat.
+
+
+Overview:
+=========
+adler32.c               adler32 checksum
+aes.c                   AES encryption and decryption
+fifo.c                  resizeable first in first out buffer
+intfloat_readwrite.c    portable reading and writing of floating point values
+log.c                   "printf" with context and level
+md5.c                   MD5 Message-Digest Algorithm
+rational.c              code to perform exact calculations with rational numbers
+tree.c                  generic AVL tree
+crc.c                   generic CRC checksumming code
+integer.c               128bit integer math
+lls.c
+mathematics.c           greatest common divisor, integer sqrt, integer log2, ...
+mem.c                   memory allocation routines with guaranteed alignment
+
+Headers:
+bswap.h                 big/little/native-endian conversion code
+x86_cpu.h               a few useful macros for unifying x86-64 and x86-32 code
+avutil.h
+common.h
+intreadwrite.h          reading and writing of unaligned big/little/native-endian integers
+
+
+Goals:
+======
+* Modular (few interdependencies and the possibility of disabling individual parts during ./configure)
+* Small (source and object)
+* Efficient (low CPU and memory usage)
+* Useful (avoid useless features almost no one needs)

doc/bitstream_filters.texi

@@ -13,16 +13,7 @@ bitstream filter using the option @code{--disable-bsf=BSF}.
 The option @code{-bsfs} of the ff* tools will display the list of
 all the supported bitstream filters included in your build.
 
-The ff* tools have a -bsf option applied per stream, taking a
-comma-separated list of filters, whose parameters follow the filter
-name after a '='.
-
-@example
-ffmpeg -i INPUT -c:v copy -bsf:v filter1[=opt1=str1/opt2=str2][,filter2] OUTPUT
-@end example
-
-Below is a description of the currently available bitstream filters,
-with their parameters, if any.
+Below is a description of the currently available bitstream filters.
 
 @section aac_adtstoasc
 
@@ -83,18 +74,7 @@ format with @command{ffmpeg}, you can use the command:
 ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts
 @end example
 
-@section imxdump
-
-Modifies the bitstream to fit in MOV and to be usable by the Final Cut
-Pro decoder. This filter only applies to the mpeg2video codec, and is
-likely not needed for Final Cut Pro 7 and newer with the appropriate
-@option{-tag:v}.
-
-For example, to remux 30 MB/sec NTSC IMX to MOV:
-
-@example
-ffmpeg -i input.mxf -c copy -bsf:v imxdump -tag:v mx3n output.mov
-@end example
+@section imx_dump_header
 
 @section mjpeg2jpeg
 
@@ -137,24 +117,12 @@ ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
 
 @section movsub
 
+@section mp3_header_compress
+
 @section mp3_header_decompress
 
 @section noise
 
-Damages the contents of packets without damaging the container. Can be
-used for fuzzing or testing error resilience/concealment.
-
-Parameters:
-A numeral string, whose value is related to how often output bytes will
-be modified. Therefore, values below or equal to 0 are forbidden, and
-the lower the more frequent bytes will be modified, with 1 meaning
-every byte is modified.
-
-@example
-ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv
-@end example
-applies the modification to every byte.
-
 @section remove_extra
 
 @c man end BITSTREAM FILTERS

doc/build_system.txt

@@ -7,11 +7,6 @@ V
     Disable the default terse mode, the full command issued by make and its
     output will be shown on the screen.
 
-DBG
-    Preprocess x86 external assembler files to a .dbg.asm file in the object
-    directory, which then gets compiled. Helps developping those assembler
-    files.
-
 DESTDIR
     Destination directory for the install targets, useful to prepare packages
     or install FFmpeg in cross-environments.
@@ -30,9 +25,6 @@ fate-list
 install
     Install headers, libraries and programs.
 
-examples
-    Build all examples located in doc/examples.
-
 libavformat/output-example
     Build the libavformat basic example.
 
@@ -42,9 +34,6 @@ libavcodec/api-example
 libswscale/swscale-test
     Build the swscale self-test (useful also as example).
 
-config
-    Reconfigure the project with current configuration.
-
 
 Useful standard make commands:
 make -t <target>

doc/codecs.texi

@@ -7,7 +7,7 @@ all the encoders and decoders. In addition each codec may support
 so-called private options, which are specific for a given codec.
 
 Sometimes, a global option may only affect a specific kind of codec,
-and may be nonsensical or ignored by another, so you need to be aware
+and may be unsensical or ignored by another, so you need to be aware
 of the meaning of the specified options. Also some options are
 meant only for decoding or encoding.
 
@@ -71,9 +71,7 @@ Force low delay.
 @item global_header
 Place global headers in extradata instead of every keyframe.
 @item bitexact
-Only write platform-, build- and time-independent data. (except (I)DCT).
-This ensures that file and data checksums are reproducible and match between
-platforms. Its primary use is for regression testing.
+Use only bitexact stuff (except (I)DCT).
 @item aic
 Apply H263 advanced intra coding / mpeg4 ac prediction.
 @item cbp
@@ -174,13 +172,7 @@ Set max video quantizer scale (VBR). Must be included between -1 and
 Set max difference between the quantizer scale (VBR).
 
 @item bf @var{integer} (@emph{encoding,video})
-Set max number of B frames between non-B-frames.
-
-Must be an integer between -1 and 16. 0 means that B-frames are
-disabled. If a value of -1 is used, it will choose an automatic value
-depending on the encoder.
-
-Default value is 0.
+Set max number of B frames.
 
 @item b_qfactor @var{float} (@emph{encoding,video})
 Set qp factor between P and B frames.
@@ -287,11 +279,6 @@ detect bitstream specification deviations
 detect improper bitstream length
 @item explode
 abort decoding on minor error detection
-@item ignore_err
-ignore decoding errors, and continue decoding.
-This is useful if you want to analyze the content of a video and thus want
-everything to be decoded no matter what. This option will not result in a video
-that is pleasing to watch in case of errors.
 @item careful
 consider things that violate the spec and have not been seen in the wild as errors
 @item compliant
@@ -396,9 +383,6 @@ Possible values:
 
 @item simplemmx
 
-@item simpleauto
-Automatically pick a IDCT compatible with the simple one
-
 @item arm
 
 @item altivec
@@ -434,8 +418,6 @@ Possible values:
 iterative motion vector (MV) search (slow)
 @item deblock
 use strong deblock filter for damaged MBs
-@item favor_inter
-favor predicting from the previous frame instead of the current
 @end table
 
 @item bits_per_coded_sample @var{integer}
@@ -495,15 +477,11 @@ visualize block types
 picture buffer allocations
 @item thread_ops
 threading operations
-@item nomc
-skip motion compensation
 @end table
 
 @item vismv @var{integer} (@emph{decoding,video})
 Visualize motion vectors (MVs).
 
-This option is deprecated, see the codecview filter instead.
-
 Possible values:
 @table @samp
 @item pf
@@ -803,9 +781,6 @@ Frame data might be split into multiple chunks.
 Show all frames before the first keyframe.
 @item skiprd
 Deprecated, use mpegvideo private options instead.
-@item export_mvs
-Export motion vectors into frame side-data (see @code{AV_FRAME_DATA_MOTION_VECTORS})
-for codecs that support it. See also @file{doc/examples/export_mvs.c}.
 @end table
 
 @item error @var{integer} (@emph{encoding,video})
@@ -865,14 +840,6 @@ Possible values:
 
 @item mpeg2_aac_he
 
-@item mpeg4_sp
-
-@item mpeg4_core
-
-@item mpeg4_main
-
-@item mpeg4_asp
-
 @item dts
 
 @item dts_es
@@ -904,9 +871,6 @@ Set frame skip factor.
 
 @item skip_exp @var{integer} (@emph{encoding,video})
 Set frame skip exponent.
-Negative values behave identical to the corresponding positive ones, except
-that the score is normalized.
-Positive values exist primarily for compatibility reasons and are not so useful.
 
 @item skipcmp @var{integer} (@emph{encoding,video})
 Set frame skip compare function.
@@ -1052,26 +1016,15 @@ Set the log level offset.
 Number of slices, used in parallelized encoding.
 
 @item thread_type @var{flags} (@emph{decoding/encoding,video})
-Select which multithreading methods to use.
-
-Use of @samp{frame} will increase decoding delay by one frame per
-thread, so clients which cannot provide future frames should not use
-it.
+Select multithreading type.
 
 Possible values:
 @table @samp
 @item slice
-Decode more than one part of a single frame at once.
-
-Multithreading using slices works only when the video was encoded with
-slices.
 
 @item frame
-Decode more than one frame at once.
+
 @end table
-
-Default value is @samp{slice+frame}.
-
 @item audio_service_type @var{integer} (@emph{encoding,audio})
 Set audio service type.
 
@@ -1126,26 +1079,9 @@ Interlaced video, bottom coded first, top displayed first
 Set to 1 to disable processing alpha (transparency). This works like the
 @samp{gray} flag in the @option{flags} option which skips chroma information
 instead of alpha. Default is 0.
-
-@item codec_whitelist @var{list} (@emph{input})
-"," separated List of allowed decoders. By default all are allowed.
-
-@item dump_separator @var{string} (@emph{input})
-Separator used to separate the fields printed on the command line about the
-Stream parameters.
-For example to separate the fields with newlines and indention:
-@example
-ffprobe -dump_separator "
-                          "  -i ~/videos/matrixbench_mpeg2.mpg
-@end example
-
 @end table
 
 @c man end CODEC OPTIONS
 
-@ifclear config-writeonly
 @include decoders.texi
-@end ifclear
-@ifclear config-readonly
 @include encoders.texi
-@end ifclear

doc/decoders.texi

@@ -14,7 +14,7 @@ You can disable all the decoders with the configure option
 with the options @code{--enable-decoder=@var{DECODER}} /
 @code{--disable-decoder=@var{DECODER}}.
 
-The option @code{-decoders} of the ff* tools will display the list of
+The option @code{-codecs} of the ff* tools will display the list of
 enabled decoders.
 
 @c man end DECODERS
@@ -52,37 +52,6 @@ top-field-first is assumed
 @chapter Audio Decoders
 @c man begin AUDIO DECODERS
 
-A description of some of the currently available audio decoders
-follows.
-
-@section ac3
-
-AC-3 audio decoder.
-
-This decoder implements part of ATSC A/52:2010 and ETSI TS 102 366, as well as
-the undocumented RealAudio 3 (a.k.a. dnet).
-
-@subsection AC-3 Decoder Options
-
-@table @option
-
-@item -drc_scale @var{value}
-Dynamic Range Scale Factor. The factor to apply to dynamic range values
-from the AC-3 stream. This factor is applied exponentially.
-There are 3 notable scale factor ranges:
-@table @option
-@item drc_scale == 0
-DRC disabled. Produces full range audio.
-@item 0 < drc_scale <= 1
-DRC enabled.  Applies a fraction of the stream DRC value.
-Audio reproduction is between full range and full compression.
-@item drc_scale > 1
-DRC enabled. Applies drc_scale asymmetrically.
-Loud sounds are fully compressed.  Soft sounds are enhanced.
-@end table
-
-@end table
-
 @section ffwavesynth
 
 Internal wave synthetizer.
@@ -163,9 +132,6 @@ Requires the presence of the libopus headers and library during
 configuration. You need to explicitly configure the build with
 @code{--enable-libopus}.
 
-An FFmpeg native decoder for Opus exists, so users can decode Opus
-without this library.
-
 @c man end AUDIO DECODERS
 
 @chapter Subtitles Decoders
@@ -190,15 +156,6 @@ The format for this option is a string containing 16 24-bits hexadecimal
 numbers (without 0x prefix) separated by comas, for example @code{0d00ee,
 ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1,
 7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b}.
-
-@item ifo_palette
-Specify the IFO file from which the global palette is obtained.
-(experimental)
-
-@item forced_subs_only
-Only decode subtitle entries marked as forced. Some titles have forced
-and non-forced subtitles in the same track. Setting this flag to @code{1}
-will only keep the forced subtitles. Default value is @code{0}.
 @end table
 
 @section libzvbi-teletext

doc/default.css

@@ -17,8 +17,8 @@ a:visited {
 }
 
 #banner img {
-    margin-bottom: 1px;
-    margin-top: 5px;
+    padding-bottom: 1px;
+    padding-top: 5px;
 }
 
 #body {

doc/demuxers.texi

@@ -29,26 +29,6 @@ the caller can decide which variant streams to actually receive.
 The total bitrate of the variant that the stream belongs to is
 available in a metadata key named "variant_bitrate".
 
-@section apng
-
-Animated Portable Network Graphics demuxer.
-
-This demuxer is used to demux APNG files.
-All headers, but the PNG signature, up to (but not including) the first
-fcTL chunk are transmitted as extradata.
-Frames are then split as being all the chunks between two fcTL ones, or
-between the last fcTL and IEND chunks.
-
-@table @option
-@item -ignore_loop @var{bool}
-Ignore the loop variable in the file if set.
-@item -max_fps @var{int}
-Maximum framerate in frames per second (0 for no limit).
-@item -default_fps @var{int}
-Default framerate in frames per second when none is specified in the file
-(0 meaning as fast as possible).
-@end table
-
 @section asf
 
 Advanced Systems Format demuxer.
@@ -94,7 +74,7 @@ following directive is recognized:
 Path to a file to read; special characters and spaces must be escaped with
 backslash or single quotes.
 
-All subsequent file-related directives apply to that file.
+All subsequent directives apply to that file.
 
 @item @code{ffconcat version 1.0}
 Identify the script type and version. It also sets the @option{safe} option
@@ -112,22 +92,6 @@ file is not available or accurate.
 If the duration is set for all files, then it is possible to seek in the
 whole concatenated video.
 
-@item @code{stream}
-Introduce a stream in the virtual file.
-All subsequent stream-related directives apply to the last introduced
-stream.
-Some streams properties must be set in order to allow identifying the
-matching streams in the subfiles.
-If no streams are defined in the script, the streams from the first file are
-copied.
-
-@item @code{exact_stream_id @var{id}}
-Set the id of the stream.
-If this directive is given, the string with the corresponding id in the
-subfiles will be used.
-This is especially useful for MPEG-PS (VOB) files, where the order of the
-streams is not reliable.
-
 @end table
 
 @subsection Options
@@ -148,14 +112,6 @@ If set to 0, any file name is accepted.
 The default is -1, it is equivalent to 1 if the format was automatically
 probed and 0 otherwise.
 
-@item auto_convert
-If set to 1, try to perform automatic conversions on packet data to make the
-streams concatenable.
-
-Currently, the only conversion is adding the h264_mp4toannexb bitstream
-filter to H.264 streams in MP4 format. This is necessary in particular if
-there are resolution changes.
-
 @end table
 
 @section flv
@@ -194,40 +150,6 @@ See @url{http://quvi.sourceforge.net/} for more information.
 FFmpeg needs to be built with @code{--enable-libquvi} for this demuxer to be
 enabled.
 
-@section gif
-
-Animated GIF demuxer.
-
-It accepts the following options:
-
-@table @option
-@item min_delay
-Set the minimum valid delay between frames in hundredths of seconds.
-Range is 0 to 6000. Default value is 2.
-
-@item default_delay
-Set the default delay between frames in hundredths of seconds.
-Range is 0 to 6000. Default value is 10.
-
-@item ignore_loop
-GIF files can contain information to loop a certain number of times (or
-infinitely). If @option{ignore_loop} is set to 1, then the loop setting
-from the input will be ignored and looping will not occur. If set to 0,
-then looping will occur and will cycle the number of times according to
-the GIF. Default value is 1.
-@end table
-
-For example, with the overlay filter, place an infinitely looping GIF
-over another video:
-@example
-ffmpeg -i input.mp4 -ignore_loop 0 -i input.gif -filter_complex overlay=shortest=1 out.mkv
-@end example
-
-Note that in the above example the shortest option for overlay filter is
-used to end the output video at the length of the shortest input file,
-which in this case is @file{input.mp4} as the GIF in this example loops
-infinitely.
-
 @section image2
 
 Image file demuxer.
@@ -327,8 +249,6 @@ is 5.
 If set to 1, will set frame timestamp to modification time of image file. Note
 that monotonity of timestamps is not provided: images go in the same order as
 without this option. Default value is 0.
-If set to 2, will set frame timestamp to the modification time of the image file in
-nanosecond precision.
 @item video_size
 Set the video size of the images to read. If not specified the video
 size is guessed from the first image file in the sequence.
@@ -376,7 +296,7 @@ teletext packet PTS and DTS values untouched.
 
 Raw video demuxer.
 
-This demuxer allows one to read raw video data. Since there is no header
+This demuxer allows to read raw video data. Since there is no header
 specifying the assumed video parameters, the user must specify them
 in order to be able to decode the data correctly.
 

doc/developer.texi

@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8
 
 @settitle Developer Documentation
 @titlepage
@@ -249,7 +248,7 @@ Contributions should be licensed under the
 @uref{http://www.gnu.org/licenses/lgpl-2.1.html, LGPL 2.1},
 including an "or any later version" clause, or, if you prefer
 a gift-style license, the
-@uref{http://opensource.org/licenses/isc-license.txt, ISC} or
+@uref{http://www.isc.org/software/license/, ISC} or
 @uref{http://mit-license.org/, MIT} license.
 @uref{http://www.gnu.org/licenses/gpl-2.0.html, GPL 2} including
 an "or any later version" clause is also acceptable, but LGPL is
@@ -324,12 +323,9 @@ Always fill out the commit log message. Describe in a few lines what you
 changed and why. You can refer to mailing list postings if you fix a
 particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
 Recommended format:
-
-@example
 area changed: Short 1 line description
 
 details describing what and why and giving references.
-@end example
 
 @item
 Make sure the author of the commit is set correctly. (see git commit --author)
@@ -648,12 +644,12 @@ accordingly].
 @subsection Adding files to the fate-suite dataset
 
 When there is no muxer or encoder available to generate test media for a
-specific test then the media has to be included in the fate-suite.
+specific test then the media has to be inlcuded in the fate-suite.
 First please make sure that the sample file is as small as possible to test the
 respective decoder or demuxer sufficiently. Large files increase network
 bandwidth and disk space requirements.
 Once you have a working fate test and fate sample, provide in the commit
-message or introductory message for the patch series that you post to
+message or introductionary message for the patch series that you post to
 the ffmpeg-devel mailing list, a direct link to download the sample media.
 
 

doc/devices.texi

@@ -17,9 +17,5 @@ for programmatic use.
 
 @c man end DEVICE OPTIONS
 
-@ifclear config-writeonly
 @include indevs.texi
-@end ifclear
-@ifclear config-readonly
 @include outdevs.texi
-@end ifclear

doc/doxy-wrapper.sh

@@ -2,20 +2,13 @@
 
 SRC_PATH="${1}"
 DOXYFILE="${2}"
-DOXYGEN="${3}"
 
-shift 3
+shift 2
 
-if [ -e "$SRC_PATH/VERSION" ]; then
-    VERSION=`cat "$SRC_PATH/VERSION"`
-else
-    VERSION=`cd "$SRC_PATH"; git describe`
-fi
-
-$DOXYGEN - <<EOF
+doxygen - <<EOF
 @INCLUDE        = ${DOXYFILE}
 INPUT           = $@
-EXAMPLE_PATH    = ${SRC_PATH}/doc/examples
-HTML_TIMESTAMP  = NO
-PROJECT_NUMBER  = $VERSION
+HTML_HEADER     = ${SRC_PATH}/doc/doxy/header.html
+HTML_FOOTER     = ${SRC_PATH}/doc/doxy/footer.html
+HTML_STYLESHEET = ${SRC_PATH}/doc/doxy/doxy_stylesheet.css
 EOF

doc/doxy/footer.html

@@ -0,0 +1,9 @@
+
+ <footer class="footer pagination-right">
+   <span class="label label-info">
+    Generated on $datetime for $projectname by&#160;<a href="http://www.doxygen.org/index.html">doxygen</a> $doxygenversion
+   </span>
+ </footer>
+</div>
+</body>
+</html>

doc/doxy/header.html

@@ -0,0 +1,16 @@
+<!DOCTYPE html>
+<html>
+<head>
+<meta http-equiv="Content-Type" content="text/html; charset=UTF-8"/>
+<meta http-equiv="X-UA-Compatible" content="IE=9"/>
+<!--BEGIN PROJECT_NAME--><title>$projectname: $title</title><!--END PROJECT_NAME-->
+<!--BEGIN !PROJECT_NAME--><title>$title</title><!--END !PROJECT_NAME-->
+<link href="$relpath$doxy_stylesheet.css" rel="stylesheet" type="text/css" />
+<!--Header replace -->
+
+</head>
+
+<div class="container">
+
+<!--Header replace -->
+<div class="menu">

doc/encoders.texi

@@ -14,7 +14,7 @@ You can disable all the encoders with the configure option
 with the options @code{--enable-encoder=@var{ENCODER}} /
 @code{--disable-encoder=@var{ENCODER}}.
 
-The option @code{-encoders} of the ff* tools will display the list of
+The option @code{-codecs} of the ff* tools will display the list of
 enabled encoders.
 
 @c man end ENCODERS
@@ -38,8 +38,8 @@ As this encoder is experimental, unexpected behavior may exist from time to
 time. For a more stable AAC encoder, see @ref{libvo-aacenc}. However, be warned
 that it has a worse quality reported by some users.
 
-@c todo @ref{libaacplus}
-See also @ref{libfdk-aac-enc,,libfdk_aac} and @ref{libfaac}.
+@c Comment this out until somebody writes the respective documentation.
+@c See also @ref{libfaac}, @ref{libaacplus}, and @ref{libfdk-aac-enc}.
 
 @subsection Options
 
@@ -80,7 +80,7 @@ thresholds with quantizer steps to find the appropriate quantization with
 distortion below threshold band by band.
 
 The quality of this method is comparable to the two loop searching method
-described below, but somewhat a little better and slower.
+descibed below, but somewhat a little better and slower.
 
 @item anmr
 Average noise to mask ratio (ANMR) trellis-based solution.
@@ -494,285 +494,6 @@ Selected by Encoder (default)
 
 @end table
 
-@anchor{libfaac}
-@section libfaac
-
-libfaac AAC (Advanced Audio Coding) encoder wrapper.
-
-Requires the presence of the libfaac headers and library during
-configuration. You need to explicitly configure the build with
-@code{--enable-libfaac --enable-nonfree}.
-
-This encoder is considered to be of higher quality with respect to the
-@ref{aacenc,,the native experimental FFmpeg AAC encoder}.
-
-For more information see the libfaac project at
-@url{http://www.audiocoding.com/faac.html/}.
-
-@subsection Options
-
-The following shared FFmpeg codec options are recognized.
-
-The following options are supported by the libfaac wrapper. The
-@command{faac}-equivalent of the options are listed in parentheses.
-
-@table @option
-@item b (@emph{-b})
-Set bit rate in bits/s for ABR (Average Bit Rate) mode. If the bit rate
-is not explicitly specified, it is automatically set to a suitable
-value depending on the selected profile. @command{faac} bitrate is
-expressed in kilobits/s.
-
-Note that libfaac does not support CBR (Constant Bit Rate) but only
-ABR (Average Bit Rate).
-
-If VBR mode is enabled this option is ignored.
-
-@item ar (@emph{-R})
-Set audio sampling rate (in Hz).
-
-@item ac (@emph{-c})
-Set the number of audio channels.
-
-@item cutoff (@emph{-C})
-Set cutoff frequency. If not specified (or explicitly set to 0) it
-will use a value automatically computed by the library. Default value
-is 0.
-
-@item profile
-Set audio profile.
-
-The following profiles are recognized:
-@table @samp
-@item aac_main
-Main AAC (Main)
-
-@item aac_low
-Low Complexity AAC (LC)
-
-@item aac_ssr
-Scalable Sample Rate (SSR)
-
-@item aac_ltp
-Long Term Prediction (LTP)
-@end table
-
-If not specified it is set to @samp{aac_low}.
-
-@item flags +qscale
-Set constant quality VBR (Variable Bit Rate) mode.
-
-@item global_quality
-Set quality in VBR mode as an integer number of lambda units.
-
-Only relevant when VBR mode is enabled with @code{flags +qscale}.  The
-value is converted to QP units by dividing it by @code{FF_QP2LAMBDA},
-and used to set the quality value used by libfaac. A reasonable range
-for the option value in QP units is [10-500], the higher the value the
-higher the quality.
-
-@item q (@emph{-q})
-Enable VBR mode when set to a non-negative value, and set constant
-quality value as a double floating point value in QP units.
-
-The value sets the quality value used by libfaac. A reasonable range
-for the option value is [10-500], the higher the value the higher the
-quality.
-
-This option is valid only using the @command{ffmpeg} command-line
-tool. For library interface users, use @option{global_quality}.
-@end table
-
-@subsection Examples
-
-@itemize
-@item
-Use @command{ffmpeg} to convert an audio file to ABR 128 kbps AAC in an M4A (MP4)
-container:
-@example
-ffmpeg -i input.wav -codec:a libfaac -b:a 128k -output.m4a
-@end example
-
-@item
-Use @command{ffmpeg} to convert an audio file to VBR AAC, using the
-LTP AAC profile:
-@example
-ffmpeg -i input.wav -c:a libfaac -profile:a aac_ltp -q:a 100 output.m4a
-@end example
-@end itemize
-
-@anchor{libfdk-aac-enc}
-@section libfdk_aac
-
-libfdk-aac AAC (Advanced Audio Coding) encoder wrapper.
-
-The libfdk-aac library is based on the Fraunhofer FDK AAC code from
-the Android project.
-
-Requires the presence of the libfdk-aac headers and library during
-configuration. You need to explicitly configure the build with
-@code{--enable-libfdk-aac}. The library is also incompatible with GPL,
-so if you allow the use of GPL, you should configure with
-@code{--enable-gpl --enable-nonfree --enable-libfdk-aac}.
-
-This encoder is considered to be of higher quality with respect to
-both @ref{aacenc,,the native experimental FFmpeg AAC encoder} and
-@ref{libfaac}.
-
-VBR encoding, enabled through the @option{vbr} or @option{flags
-+qscale} options, is experimental and only works with some
-combinations of parameters.
-
-Support for encoding 7.1 audio is only available with libfdk-aac 0.1.3 or
-higher.
-
-For more information see the fdk-aac project at
-@url{http://sourceforge.net/p/opencore-amr/fdk-aac/}.
-
-@subsection Options
-
-The following options are mapped on the shared FFmpeg codec options.
-
-@table @option
-@item b
-Set bit rate in bits/s. If the bitrate is not explicitly specified, it
-is automatically set to a suitable value depending on the selected
-profile.
-
-In case VBR mode is enabled the option is ignored.
-
-@item ar
-Set audio sampling rate (in Hz).
-
-@item channels
-Set the number of audio channels.
-
-@item flags +qscale
-Enable fixed quality, VBR (Variable Bit Rate) mode.
-Note that VBR is implicitly enabled when the @option{vbr} value is
-positive.
-
-@item cutoff
-Set cutoff frequency. If not specified (or explicitly set to 0) it
-will use a value automatically computed by the library. Default value
-is 0.
-
-@item profile
-Set audio profile.
-
-The following profiles are recognized:
-@table @samp
-@item aac_low
-Low Complexity AAC (LC)
-
-@item aac_he
-High Efficiency AAC (HE-AAC)
-
-@item aac_he_v2
-High Efficiency AAC version 2 (HE-AACv2)
-
-@item aac_ld
-Low Delay AAC (LD)
-
-@item aac_eld
-Enhanced Low Delay AAC (ELD)
-@end table
-
-If not specified it is set to @samp{aac_low}.
-@end table
-
-The following are private options of the libfdk_aac encoder.
-
-@table @option
-@item afterburner
-Enable afterburner feature if set to 1, disabled if set to 0. This
-improves the quality but also the required processing power.
-
-Default value is 1.
-
-@item eld_sbr
-Enable SBR (Spectral Band Replication) for ELD if set to 1, disabled
-if set to 0.
-
-Default value is 0.
-
-@item signaling
-Set SBR/PS signaling style.
-
-It can assume one of the following values:
-@table @samp
-@item default
-choose signaling implicitly (explicit hierarchical by default,
-implicit if global header is disabled)
-
-@item implicit
-implicit backwards compatible signaling
-
-@item explicit_sbr
-explicit SBR, implicit PS signaling
-
-@item explicit_hierarchical
-explicit hierarchical signaling
-@end table
-
-Default value is @samp{default}.
-
-@item latm
-Output LATM/LOAS encapsulated data if set to 1, disabled if set to 0.
-
-Default value is 0.
-
-@item header_period
-Set StreamMuxConfig and PCE repetition period (in frames) for sending
-in-band configuration buffers within LATM/LOAS transport layer.
-
-Must be a 16-bits non-negative integer.
-
-Default value is 0.
-
-@item vbr
-Set VBR mode, from 1 to 5. 1 is lowest quality (though still pretty
-good) and 5 is highest quality. A value of 0 will disable VBR, and CBR
-(Constant Bit Rate) is enabled.
-
-Currently only the @samp{aac_low} profile supports VBR encoding.
-
-VBR modes 1-5 correspond to roughly the following average bit rates:
-
-@table @samp
-@item 1
-32 kbps/channel
-@item 2
-40 kbps/channel
-@item 3
-48-56 kbps/channel
-@item 4
-64 kbps/channel
-@item 5
-about 80-96 kbps/channel
-@end table
-
-Default value is 0.
-@end table
-
-@subsection Examples
-
-@itemize
-@item
-Use @command{ffmpeg} to convert an audio file to VBR AAC in an M4A (MP4)
-container:
-@example
-ffmpeg -i input.wav -codec:a libfdk_aac -vbr 3 output.m4a
-@end example
-
-@item
-Use @command{ffmpeg} to convert an audio file to CBR 64k kbps AAC, using the
-High-Efficiency AAC profile:
-@example
-ffmpeg -i input.wav -c:a libfdk_aac -profile:a aac_he -b:a 64k output.m4a
-@end example
-@end itemize
-
 @anchor{libmp3lame}
 @section libmp3lame
 
@@ -792,7 +513,7 @@ The following options are supported by the libmp3lame wrapper. The
 
 @table @option
 @item b (@emph{-b})
-Set bitrate expressed in bits/s for CBR or ABR. LAME @code{bitrate} is
+Set bitrate expressed in bits/s for CBR. LAME @code{bitrate} is
 expressed in kilobits/s.
 
 @item q (@emph{-V})
@@ -807,18 +528,13 @@ while producing the worst quality.
 
 @item reservoir
 Enable use of bit reservoir when set to 1. Default value is 1. LAME
-has this enabled by default, but can be overridden by use
+has this enabled by default, but can be overriden by use
 @option{--nores} option.
 
 @item joint_stereo (@emph{-m j})
 Enable the encoder to use (on a frame by frame basis) either L/R
 stereo or mid/side stereo. Default value is 1.
 
-@item abr (@emph{--abr})
-Enable the encoder to use ABR when set to 1. The @command{lame}
-@option{--abr} sets the target bitrate, while this options only
-tells FFmpeg to use ABR still relies on @option{b} to set bitrate.
-
 @end table
 
 @section libopencore-amrnb
@@ -1032,7 +748,7 @@ configuration. You need to explicitly configure the build with
 
 @subsection Option Mapping
 
-Most libopus options are modelled after the @command{opusenc} utility from
+Most libopus options are modeled after the @command{opusenc} utility from
 opus-tools. The following is an option mapping chart describing options
 supported by the libopus wrapper, and their @command{opusenc}-equivalent
 in parentheses.
@@ -1147,111 +863,32 @@ transient response is a higher bitrate.
 
 @end table
 
-@anchor{libwavpack}
 @section libwavpack
 
 A wrapper providing WavPack encoding through libwavpack.
 
 Only lossless mode using 32-bit integer samples is supported currently.
-
-Requires the presence of the libwavpack headers and library during
-configuration. You need to explicitly configure the build with
-@code{--enable-libwavpack}.
-
-Note that a libavcodec-native encoder for the WavPack codec exists so users can
-encode audios with this codec without using this encoder. See @ref{wavpackenc}.
-
-@subsection Options
-
-@command{wavpack} command line utility's corresponding options are listed in
-parentheses, if any.
+The @option{compression_level} option can be used to control speed vs.
+compression tradeoff, with the values mapped to libwavpack as follows:
 
 @table @option
-@item frame_size (@emph{--blocksize})
-Default is 32768.
 
-@item compression_level
-Set speed vs. compression tradeoff. Acceptable arguments are listed below:
-
-@table @samp
-@item 0 (@emph{-f})
-Fast mode.
+@item 0
+Fast mode - corresponding to the wavpack @option{-f} option.
 
 @item 1
 Normal (default) settings.
 
-@item 2 (@emph{-h})
-High quality.
+@item 2
+High quality - corresponding to the wavpack @option{-h} option.
 
-@item 3 (@emph{-hh})
-Very high quality.
+@item 3
+Very high quality - corresponding to the wavpack @option{-hh} option.
 
-@item 4-8 (@emph{-hh -x}@var{EXTRAPROC})
-Same as @samp{3}, but with extra processing enabled.
-
-@samp{4} is the same as @option{-x2} and @samp{8} is the same as @option{-x6}.
-
-@end table
-@end table
-
-@anchor{wavpackenc}
-@section wavpack
-
-WavPack lossless audio encoder.
-
-This is a libavcodec-native WavPack encoder. There is also an encoder based on
-libwavpack, but there is virtually no reason to use that encoder.
-
-See also @ref{libwavpack}.
-
-@subsection Options
-
-The equivalent options for @command{wavpack} command line utility are listed in
-parentheses.
-
-@subsubsection Shared options
-
-The following shared options are effective for this encoder. Only special notes
-about this particular encoder will be documented here. For the general meaning
-of the options, see @ref{codec-options,,the Codec Options chapter}.
-
-@table @option
-@item frame_size (@emph{--blocksize})
-For this encoder, the range for this option is between 128 and 131072. Default
-is automatically decided based on sample rate and number of channel.
-
-For the complete formula of calculating default, see
-@file{libavcodec/wavpackenc.c}.
-
-@item compression_level (@emph{-f}, @emph{-h}, @emph{-hh}, and @emph{-x})
-This option's syntax is consistent with @ref{libwavpack}'s.
-@end table
-
-@subsubsection Private options
-
-@table @option
-@item joint_stereo (@emph{-j})
-Set whether to enable joint stereo. Valid values are:
-
-@table @samp
-@item on (@emph{1})
-Force mid/side audio encoding.
-@item off (@emph{0})
-Force left/right audio encoding.
-@item auto
-Let the encoder decide automatically.
-@end table
-
-@item optimize_mono
-Set whether to enable optimization for mono. This option is only effective for
-non-mono streams. Available values:
-
-@table @samp
-@item on
-enabled
-@item off
-disabled
-@end table
+@item 4-8
+Same as 3, but with extra processing enabled - corresponding to the wavpack
+@option{-x} option. I.e. 4 is the same as @option{-x2} and 8 is the same as
+@option{-x6}.
 
 @end table
 
@@ -1265,15 +902,12 @@ follows.
 
 @section libtheora
 
-libtheora Theora encoder wrapper.
+Theora format supported through libtheora.
 
 Requires the presence of the libtheora headers and library during
 configuration. You need to explicitly configure the build with
 @code{--enable-libtheora}.
 
-For more information about the libtheora project see
-@url{http://www.theora.org/}.
-
 @subsection Options
 
 The following global options are mapped to internal libtheora options
@@ -1281,11 +915,11 @@ which affect the quality and the bitrate of the encoded stream.
 
 @table @option
 @item b
-Set the video bitrate in bit/s for CBR (Constant Bit Rate) mode.  In
-case VBR (Variable Bit Rate) mode is enabled this option is ignored.
+Set the video bitrate, only works if the @code{qscale} flag in
+@option{flags} is not enabled.
 
 @item flags
-Used to enable constant quality mode (VBR) encoding through the
+Used to enable constant quality mode encoding through the
 @option{qscale} flag, and to enable the @code{pass1} and @code{pass2}
 modes.
 
@@ -1293,44 +927,22 @@ modes.
 Set the GOP size.
 
 @item global_quality
-Set the global quality as an integer in lambda units.
+Set the global quality in lambda units, only works if the
+@code{qscale} flag in @option{flags} is enabled. The value is clipped
+in the [0 - 10*@code{FF_QP2LAMBDA}] range, and then multiplied for 6.3
+to get a value in the native libtheora range [0-63]. A higher value
+corresponds to a higher quality.
 
-Only relevant when VBR mode is enabled with @code{flags +qscale}. The
-value is converted to QP units by dividing it by @code{FF_QP2LAMBDA},
-clipped in the [0 - 10] range, and then multiplied by 6.3 to get a
-value in the native libtheora range [0-63]. A higher value corresponds
-to a higher quality.
-
-@item q
-Enable VBR mode when set to a non-negative value, and set constant
-quality value as a double floating point value in QP units.
-
-The value is clipped in the [0-10] range, and then multiplied by 6.3
-to get a value in the native libtheora range [0-63].
-
-This option is valid only using the @command{ffmpeg} command-line
-tool. For library interface users, use @option{global_quality}.
+For example, to set maximum constant quality encoding with
+@command{ffmpeg}:
+@example
+ffmpeg -i INPUT -flags:v qscale -global_quality:v "10*QP2LAMBDA" -codec:v libtheora OUTPUT.ogg
+@end example
 @end table
 
-@subsection Examples
-
-@itemize
-@item
-Set maximum constant quality (VBR) encoding with @command{ffmpeg}:
-@example
-ffmpeg -i INPUT -codec:v libtheora -q:v 10 OUTPUT.ogg
-@end example
-
-@item
-Use @command{ffmpeg} to convert a CBR 1000 kbps Theora video stream:
-@example
-ffmpeg -i INPUT -codec:v libtheora -b:v 1000k OUTPUT.ogg
-@end example
-@end itemize
-
 @section libvpx
 
-VP8/VP9 format supported through libvpx.
+VP8 format supported through libvpx.
 
 Requires the presence of the libvpx headers and library during configuration.
 You need to explicitly configure the build with @code{--enable-libvpx}.
@@ -1442,77 +1054,12 @@ g_lag_in_frames
 @item vp8flags error_resilient
 g_error_resilient
 
-@item aq_mode
-@code{VP9E_SET_AQ_MODE}
-
 @end table
 
 For more information about libvpx see:
 @url{http://www.webmproject.org/}
 
-
-@section libwebp
-
-libwebp WebP Image encoder wrapper
-
-libwebp is Google's official encoder for WebP images. It can encode in either
-lossy or lossless mode. Lossy images are essentially a wrapper around a VP8
-frame. Lossless images are a separate codec developed by Google.
-
-@subsection Pixel Format
-
-Currently, libwebp only supports YUV420 for lossy and RGB for lossless due
-to limitations of the format and libwebp. Alpha is supported for either mode.
-Because of API limitations, if RGB is passed in when encoding lossy or YUV is
-passed in for encoding lossless, the pixel format will automatically be
-converted using functions from libwebp. This is not ideal and is done only for
-convenience.
-
-@subsection Options
-
-@table @option
-
-@item -lossless @var{boolean}
-Enables/Disables use of lossless mode. Default is 0.
-
-@item -compression_level @var{integer}
-For lossy, this is a quality/speed tradeoff. Higher values give better quality
-for a given size at the cost of increased encoding time. For lossless, this is
-a size/speed tradeoff. Higher values give smaller size at the cost of increased
-encoding time. More specifically, it controls the number of extra algorithms
-and compression tools used, and varies the combination of these tools. This
-maps to the @var{method} option in libwebp. The valid range is 0 to 6.
-Default is 4.
-
-@item -qscale @var{float}
-For lossy encoding, this controls image quality, 0 to 100. For lossless
-encoding, this controls the effort and time spent at compressing more. The
-default value is 75. Note that for usage via libavcodec, this option is called
-@var{global_quality} and must be multiplied by @var{FF_QP2LAMBDA}.
-
-@item -preset @var{type}
-Configuration preset. This does some automatic settings based on the general
-type of the image.
-@table @option
-@item none
-Do not use a preset.
-@item default
-Use the encoder default.
-@item picture
-Digital picture, like portrait, inner shot
-@item photo
-Outdoor photograph, with natural lighting
-@item drawing
-Hand or line drawing, with high-contrast details
-@item icon
-Small-sized colorful images
-@item text
-Text-like
-@end table
-
-@end table
-
-@section libx264, libx264rgb
+@section libx264
 
 x264 H.264/MPEG-4 AVC encoder wrapper.
 
@@ -1528,22 +1075,12 @@ for detail retention (adaptive quantization, psy-RD, psy-trellis).
 Many libx264 encoder options are mapped to FFmpeg global codec
 options, while unique encoder options are provided through private
 options. Additionally the @option{x264opts} and @option{x264-params}
-private options allows one to pass a list of key=value tuples as accepted
+private options allows to pass a list of key=value tuples as accepted
 by the libx264 @code{x264_param_parse} function.
 
 The x264 project website is at
 @url{http://www.videolan.org/developers/x264.html}.
 
-The libx264rgb encoder is the same as libx264, except it accepts packed RGB
-pixel formats as input instead of YUV.
-
-@subsection Supported Pixel Formats
-
-x264 supports 8- to 10-bit color spaces. The exact bit depth is controlled at
-x264's configure time. FFmpeg only supports one bit depth in one particular
-build. In other words, it is not possible to build one FFmpeg with multiple
-versions of x264 with different bit depths.
-
 @subsection Options
 
 The following options are supported by the libx264 wrapper. The
@@ -1569,34 +1106,25 @@ kilobits/s.
 
 @item g (@emph{keyint})
 
-@item qmin (@emph{qpmin})
-Minimum quantizer scale.
-
 @item qmax (@emph{qpmax})
-Maximum quantizer scale.
+
+@item qmin (@emph{qpmin})
 
 @item qdiff (@emph{qpstep})
-Maximum difference between quantizer scales.
 
 @item qblur (@emph{qblur})
-Quantizer curve blur
 
 @item qcomp (@emph{qcomp})
-Quantizer curve compression factor
 
 @item refs (@emph{ref})
-Number of reference frames each P-frame can use. The range is from @var{0-16}.
 
 @item sc_threshold (@emph{scenecut})
-Sets the threshold for the scene change detection.
 
 @item trellis (@emph{trellis})
-Performs Trellis quantization to increase efficiency. Enabled by default.
 
 @item nr  (@emph{nr})
 
 @item me_range (@emph{merange})
-Maximum range of the motion search in pixels.
 
 @item me_method (@emph{me})
 Set motion estimation method. Possible values in the decreasing order
@@ -1618,13 +1146,10 @@ Hadamard exhaustive search (slowest).
 @end table
 
 @item subq (@emph{subme})
-Sub-pixel motion estimation method.
 
 @item b_strategy (@emph{b-adapt})
-Adaptive B-frame placement decision algorithm. Use only on first-pass.
 
 @item keyint_min (@emph{min-keyint})
-Minimum GOP size.
 
 @item coder
 Set entropy encoder. Possible values:
@@ -1651,7 +1176,6 @@ Ignore chroma in motion estimation. It generates the same effect as
 @end table
 
 @item threads (@emph{threads})
-Number of encoding threads.
 
 @item thread_type
 Set multithreading technique. Possible values:
@@ -1745,10 +1269,6 @@ Enable calculation and printing SSIM stats after the encoding.
 Enable the use of Periodic Intra Refresh instead of IDR frames when set
 to 1.
 
-@item avcintra-class (@emph{class})
-Configure the encoder to generate AVC-Intra.
-Valid values are 50,100 and 200
-
 @item bluray-compat (@emph{bluray-compat})
 Configure the encoder to be compatible with the bluray standard.
 It is a shorthand for setting "bluray-compat=1 force-cfr=1".
@@ -1873,7 +1393,7 @@ Override the x264 configuration using a :-separated list of key=value
 parameters.
 
 This option is functionally the same as the @option{x264opts}, but is
-duplicated for compatibility with the Libav fork.
+duplicated for compability with the Libav fork.
 
 For example to specify libx264 encoding options with @command{ffmpeg}:
 @example
@@ -1886,34 +1406,6 @@ no-fast-pskip=1:subq=6:8x8dct=0:trellis=0 OUTPUT
 Encoding ffpresets for common usages are provided so they can be used with the
 general presets system (e.g. passing the @option{pre} option).
 
-@section libx265
-
-x265 H.265/HEVC encoder wrapper.
-
-This encoder requires the presence of the libx265 headers and library
-during configuration. You need to explicitly configure the build with
-@option{--enable-libx265}.
-
-@subsection Options
-
-@table @option
-@item preset
-Set the x265 preset.
-
-@item tune
-Set the x265 tune parameter.
-
-@item x265-params
-Set x265 options using a list of @var{key}=@var{value} couples separated
-by ":". See @command{x265 --help} for a list of options.
-
-For example to specify libx265 encoding options with @option{-x265-params}:
-
-@example
-ffmpeg -i input -c:v libx265 -x265-params crf=26:psy-rd=1 output.mp4
-@end example
-@end table
-
 @section libxvid
 
 Xvid MPEG-4 Part 2 encoder wrapper.
@@ -2077,30 +1569,6 @@ fastest.
 
 @end table
 
-@section mpeg2
-
-MPEG-2 video encoder.
-
-@subsection Options
-
-@table @option
-@item seq_disp_ext @var{integer}
-Specifies if the encoder should write a sequence_display_extension to the
-output.
-@table @option
-@item -1
-@itemx auto
-Decide automatically to write it or not (this is the default) by checking if
-the data to be written is different from the default or unspecified values.
-@item 0
-@itemx never
-Never write it.
-@item 1
-@itemx always
-Always write it.
-@end table
-@end table
-
 @section png
 
 PNG image encoder.
@@ -2119,7 +1587,7 @@ Set physical density of pixels, in dots per meter, unset by default
 Apple ProRes encoder.
 
 FFmpeg contains 2 ProRes encoders, the prores-aw and prores-ks encoder.
-The used encoder can be chosen with the @code{-vcodec} option.
+The used encoder can be choosen with the @code{-vcodec} option.
 
 @subsection Private Options for prores-ks
 
@@ -2171,7 +1639,7 @@ Use @var{0} to disable alpha plane coding.
 @subsection Speed considerations
 
 In the default mode of operation the encoder has to honor frame constraints
-(i.e. not produce frames with size bigger than requested) while still making
+(i.e. not produc frames with size bigger than requested) while still making
 output picture as good as possible.
 A frame containing a lot of small details is harder to compress and the encoder
 would spend more time searching for appropriate quantizers for each slice.
@@ -2182,27 +1650,3 @@ For the fastest encoding speed set the @option{qscale} parameter (4 is the
 recommended value) and do not set a size constraint.
 
 @c man end VIDEO ENCODERS
-
-@chapter Subtitles Encoders
-@c man begin SUBTITLES ENCODERS
-
-@section dvdsub
-
-This codec encodes the bitmap subtitle format that is used in DVDs.
-Typically they are stored in VOBSUB file pairs (*.idx + *.sub),
-and they can also be used in Matroska files.
-
-@subsection Options
-
-@table @option
-@item even_rows_fix
-When set to 1, enable a work-around that makes the number of pixel rows
-even in all subtitles.  This fixes a problem with some players that
-cut off the bottom row if the number is odd.  The work-around just adds
-a fully transparent row if needed.  The overhead is low, typically
-one byte per subtitle on average.
-
-By default, this work-around is disabled.
-@end table
-
-@c man end SUBTITLES ENCODERS

doc/examples/Makefile

@@ -11,24 +11,18 @@ CFLAGS += -Wall -g
 CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
 LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
 
-EXAMPLES=       avio_reading                       \
-                decoding_encoding                  \
-                demuxing_decoding                  \
-                extract_mvs                        \
+EXAMPLES=       decoding_encoding                  \
+                demuxing                           \
                 filtering_video                    \
                 filtering_audio                    \
                 metadata                           \
                 muxing                             \
-                remuxing                           \
                 resampling_audio                   \
                 scaling_video                      \
-                transcode_aac                      \
-                transcoding                        \
 
 OBJS=$(addsuffix .o,$(EXAMPLES))
 
 # the following examples make explicit use of the math library
-avcodec:           LDLIBS += -lm
 decoding_encoding: LDLIBS += -lm
 muxing:            LDLIBS += -lm
 resampling_audio:  LDLIBS += -lm

doc/examples/README

@@ -5,19 +5,14 @@ Both following use cases rely on pkg-config and make, thus make sure
 that you have them installed and working on your system.
 
 
-Method 1: build the installed examples in a generic read/write user directory
+1) Build the installed examples in a generic read/write user directory
 
 Copy to a read/write user directory and just use "make", it will link
 to the libraries on your system, assuming the PKG_CONFIG_PATH is
 correctly configured.
 
-Method 2: build the examples in-tree
+2) Build the examples in-tree
 
 Assuming you are in the source FFmpeg checkout directory, you need to build
-FFmpeg (no need to make install in any prefix). Then just run "make examples".
-This will build the examples using the FFmpeg build system. You can clean those
-examples using "make examplesclean"
-
-If you want to try the dedicated Makefile examples (to emulate the first
-method), go into doc/examples and run a command such as
-PKG_CONFIG_PATH=pc-uninstalled make.
+FFmpeg (no need to make install in any prefix). Then you can go into
+doc/examples and run a command such as PKG_CONFIG_PATH=pc-uninstalled make.

doc/examples/avio_reading.c

@@ -1,134 +0,0 @@
-/*
- * Copyright (c) 2014 Stefano Sabatini
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
- * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- * THE SOFTWARE.
- */
-
-/**
- * @file
- * libavformat AVIOContext API example.
- *
- * Make libavformat demuxer access media content through a custom
- * AVIOContext read callback.
- * @example avio_reading.c
- */
-
-#include <libavcodec/avcodec.h>
-#include <libavformat/avformat.h>
-#include <libavformat/avio.h>
-#include <libavutil/file.h>
-
-struct buffer_data {
-    uint8_t *ptr;
-    size_t size; ///< size left in the buffer
-};
-
-static int read_packet(void *opaque, uint8_t *buf, int buf_size)
-{
-    struct buffer_data *bd = (struct buffer_data *)opaque;
-    buf_size = FFMIN(buf_size, bd->size);
-
-    printf("ptr:%p size:%zu\n", bd->ptr, bd->size);
-
-    /* copy internal buffer data to buf */
-    memcpy(buf, bd->ptr, buf_size);
-    bd->ptr  += buf_size;
-    bd->size -= buf_size;
-
-    return buf_size;
-}
-
-int main(int argc, char *argv[])
-{
-    AVFormatContext *fmt_ctx = NULL;
-    AVIOContext *avio_ctx = NULL;
-    uint8_t *buffer = NULL, *avio_ctx_buffer = NULL;
-    size_t buffer_size, avio_ctx_buffer_size = 4096;
-    char *input_filename = NULL;
-    int ret = 0;
-    struct buffer_data bd = { 0 };
-
-    if (argc != 2) {
-        fprintf(stderr, "usage: %s input_file\n"
-                "API example program to show how to read from a custom buffer "
-                "accessed through AVIOContext.\n", argv[0]);
-        return 1;
-    }
-    input_filename = argv[1];
-
-    /* register codecs and formats and other lavf/lavc components*/
-    av_register_all();
-
-    /* slurp file content into buffer */
-    ret = av_file_map(input_filename, &buffer, &buffer_size, 0, NULL);
-    if (ret < 0)
-        goto end;
-
-    /* fill opaque structure used by the AVIOContext read callback */
-    bd.ptr  = buffer;
-    bd.size = buffer_size;
-
-    if (!(fmt_ctx = avformat_alloc_context())) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-
-    avio_ctx_buffer = av_malloc(avio_ctx_buffer_size);
-    if (!avio_ctx_buffer) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-    avio_ctx = avio_alloc_context(avio_ctx_buffer, avio_ctx_buffer_size,
-                                  0, &bd, &read_packet, NULL, NULL);
-    if (!avio_ctx) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-    fmt_ctx->pb = avio_ctx;
-
-    ret = avformat_open_input(&fmt_ctx, NULL, NULL, NULL);
-    if (ret < 0) {
-        fprintf(stderr, "Could not open input\n");
-        goto end;
-    }
-
-    ret = avformat_find_stream_info(fmt_ctx, NULL);
-    if (ret < 0) {
-        fprintf(stderr, "Could not find stream information\n");
-        goto end;
-    }
-
-    av_dump_format(fmt_ctx, 0, input_filename, 0);
-
-end:
-    avformat_close_input(&fmt_ctx);
-    /* note: the internal buffer could have changed, and be != avio_ctx_buffer */
-    if (avio_ctx) {
-        av_freep(&avio_ctx->buffer);
-        av_freep(&avio_ctx);
-    }
-    av_file_unmap(buffer, buffer_size);
-
-    if (ret < 0) {
-        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
-        return 1;
-    }
-
-    return 0;
-}

doc/examples/decoding_encoding.c

@@ -24,10 +24,10 @@
  * @file
  * libavcodec API use example.
  *
- * @example decoding_encoding.c
  * Note that libavcodec only handles codecs (mpeg, mpeg4, etc...),
  * not file formats (avi, vob, mp4, mov, mkv, mxf, flv, mpegts, mpegps, etc...). See library 'libavformat' for the
  * format handling
+ * @example doc/examples/decoding_encoding.c
  */
 
 #include <math.h>
@@ -156,7 +156,7 @@ static void audio_encode_example(const char *filename)
     }
 
     /* frame containing input raw audio */
-    frame = av_frame_alloc();
+    frame = avcodec_alloc_frame();
     if (!frame) {
         fprintf(stderr, "Could not allocate audio frame\n");
         exit(1);
@@ -170,10 +170,6 @@ static void audio_encode_example(const char *filename)
      * we calculate the size of the samples buffer in bytes */
     buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
                                              c->sample_fmt, 0);
-    if (buffer_size < 0) {
-        fprintf(stderr, "Could not get sample buffer size\n");
-        exit(1);
-    }
     samples = av_malloc(buffer_size);
     if (!samples) {
         fprintf(stderr, "Could not allocate %d bytes for samples buffer\n",
@@ -231,7 +227,7 @@ static void audio_encode_example(const char *filename)
     fclose(f);
 
     av_freep(&samples);
-    av_frame_free(&frame);
+    avcodec_free_frame(&frame);
     avcodec_close(c);
     av_free(c);
 }
@@ -288,15 +284,15 @@ static void audio_decode_example(const char *outfilename, const char *filename)
     avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
 
     while (avpkt.size > 0) {
-        int i, ch;
         int got_frame = 0;
 
         if (!decoded_frame) {
-            if (!(decoded_frame = av_frame_alloc())) {
+            if (!(decoded_frame = avcodec_alloc_frame())) {
                 fprintf(stderr, "Could not allocate audio frame\n");
                 exit(1);
             }
-        }
+        } else
+            avcodec_get_frame_defaults(decoded_frame);
 
         len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
         if (len < 0) {
@@ -305,15 +301,10 @@ static void audio_decode_example(const char *outfilename, const char *filename)
         }
         if (got_frame) {
             /* if a frame has been decoded, output it */
-            int data_size = av_get_bytes_per_sample(c->sample_fmt);
-            if (data_size < 0) {
-                /* This should not occur, checking just for paranoia */
-                fprintf(stderr, "Failed to calculate data size\n");
-                exit(1);
-            }
-            for (i=0; i<decoded_frame->nb_samples; i++)
-                for (ch=0; ch<c->channels; ch++)
-                    fwrite(decoded_frame->data[ch] + data_size*i, 1, data_size, outfile);
+            int data_size = av_samples_get_buffer_size(NULL, c->channels,
+                                                       decoded_frame->nb_samples,
+                                                       c->sample_fmt, 1);
+            fwrite(decoded_frame->data[0], 1, data_size, outfile);
         }
         avpkt.size -= len;
         avpkt.data += len;
@@ -338,7 +329,7 @@ static void audio_decode_example(const char *outfilename, const char *filename)
 
     avcodec_close(c);
     av_free(c);
-    av_frame_free(&decoded_frame);
+    avcodec_free_frame(&decoded_frame);
 }
 
 /*
@@ -376,13 +367,7 @@ static void video_encode_example(const char *filename, int codec_id)
     c->height = 288;
     /* frames per second */
     c->time_base= (AVRational){1,25};
-    /* emit one intra frame every ten frames
-     * check frame pict_type before passing frame
-     * to encoder, if frame->pict_type is AV_PICTURE_TYPE_I
-     * then gop_size is ignored and the output of encoder
-     * will always be I frame irrespective to gop_size
-     */
-    c->gop_size = 10;
+    c->gop_size = 10; /* emit one intra frame every ten frames */
     c->max_b_frames=1;
     c->pix_fmt = AV_PIX_FMT_YUV420P;
 
@@ -401,7 +386,7 @@ static void video_encode_example(const char *filename, int codec_id)
         exit(1);
     }
 
-    frame = av_frame_alloc();
+    frame = avcodec_alloc_frame();
     if (!frame) {
         fprintf(stderr, "Could not allocate video frame\n");
         exit(1);
@@ -482,7 +467,7 @@ static void video_encode_example(const char *filename, int codec_id)
     avcodec_close(c);
     av_free(c);
     av_freep(&frame->data[0]);
-    av_frame_free(&frame);
+    avcodec_free_frame(&frame);
     printf("\n");
 }
 
@@ -580,7 +565,7 @@ static void video_decode_example(const char *outfilename, const char *filename)
         exit(1);
     }
 
-    frame = av_frame_alloc();
+    frame = avcodec_alloc_frame();
     if (!frame) {
         fprintf(stderr, "Could not allocate video frame\n");
         exit(1);
@@ -624,7 +609,7 @@ static void video_decode_example(const char *outfilename, const char *filename)
 
     avcodec_close(c);
     av_free(c);
-    av_frame_free(&frame);
+    avcodec_free_frame(&frame);
     printf("\n");
 }
 
@@ -641,7 +626,7 @@ int main(int argc, char **argv)
                "This program generates a synthetic stream and encodes it to a file\n"
                "named test.h264, test.mp2 or test.mpg depending on output_type.\n"
                "The encoded stream is then decoded and written to a raw data output.\n"
-               "output_type must be chosen between 'h264', 'mp2', 'mpg'.\n",
+               "output_type must be choosen between 'h264', 'mp2', 'mpg'.\n",
                argv[0]);
         return 1;
     }
@@ -651,7 +636,7 @@ int main(int argc, char **argv)
         video_encode_example("test.h264", AV_CODEC_ID_H264);
     } else if (!strcmp(output_type, "mp2")) {
         audio_encode_example("test.mp2");
-        audio_decode_example("test.pcm", "test.mp2");
+        audio_decode_example("test.sw", "test.mp2");
     } else if (!strcmp(output_type, "mpg")) {
         video_encode_example("test.mpg", AV_CODEC_ID_MPEG1VIDEO);
         video_decode_example("test%02d.pgm", "test.mpg");

doc/examples/demuxing.c

@@ -22,11 +22,11 @@
 
 /**
  * @file
- * Demuxing and decoding example.
+ * libavformat demuxing API use example.
  *
  * Show how to use the libavformat and libavcodec API to demux and
  * decode audio and video data.
- * @example demuxing_decoding.c
+ * @example doc/examples/demuxing.c
  */
 
 #include <libavutil/imgutils.h>
@@ -36,8 +36,6 @@
 
 static AVFormatContext *fmt_ctx = NULL;
 static AVCodecContext *video_dec_ctx = NULL, *audio_dec_ctx;
-static int width, height;
-static enum AVPixelFormat pix_fmt;
 static AVStream *video_stream = NULL, *audio_stream = NULL;
 static const char *src_filename = NULL;
 static const char *video_dst_filename = NULL;
@@ -55,50 +53,20 @@ static AVPacket pkt;
 static int video_frame_count = 0;
 static int audio_frame_count = 0;
 
-/* The different ways of decoding and managing data memory. You are not
- * supposed to support all the modes in your application but pick the one most
- * appropriate to your needs. Look for the use of api_mode in this example to
- * see what are the differences of API usage between them */
-enum {
-    API_MODE_OLD                  = 0, /* old method, deprecated */
-    API_MODE_NEW_API_REF_COUNT    = 1, /* new method, using the frame reference counting */
-    API_MODE_NEW_API_NO_REF_COUNT = 2, /* new method, without reference counting */
-};
-
-static int api_mode = API_MODE_OLD;
-
 static int decode_packet(int *got_frame, int cached)
 {
     int ret = 0;
     int decoded = pkt.size;
 
-    *got_frame = 0;
-
     if (pkt.stream_index == video_stream_idx) {
         /* decode video frame */
         ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
         if (ret < 0) {
-            fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
+            fprintf(stderr, "Error decoding video frame\n");
             return ret;
         }
 
         if (*got_frame) {
-
-            if (frame->width != width || frame->height != height ||
-                frame->format != pix_fmt) {
-                /* To handle this change, one could call av_image_alloc again and
-                 * decode the following frames into another rawvideo file. */
-                fprintf(stderr, "Error: Width, height and pixel format have to be "
-                        "constant in a rawvideo file, but the width, height or "
-                        "pixel format of the input video changed:\n"
-                        "old: width = %d, height = %d, format = %s\n"
-                        "new: width = %d, height = %d, format = %s\n",
-                        width, height, av_get_pix_fmt_name(pix_fmt),
-                        frame->width, frame->height,
-                        av_get_pix_fmt_name(frame->format));
-                return -1;
-            }
-
             printf("video_frame%s n:%d coded_n:%d pts:%s\n",
                    cached ? "(cached)" : "",
                    video_frame_count++, frame->coded_picture_number,
@@ -108,7 +76,7 @@ static int decode_packet(int *got_frame, int cached)
              * this is required since rawvideo expects non aligned data */
             av_image_copy(video_dst_data, video_dst_linesize,
                           (const uint8_t **)(frame->data), frame->linesize,
-                          pix_fmt, width, height);
+                          video_dec_ctx->pix_fmt, video_dec_ctx->width, video_dec_ctx->height);
 
             /* write to rawvideo file */
             fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file);
@@ -117,7 +85,7 @@ static int decode_packet(int *got_frame, int cached)
         /* decode audio frame */
         ret = avcodec_decode_audio4(audio_dec_ctx, frame, got_frame, &pkt);
         if (ret < 0) {
-            fprintf(stderr, "Error decoding audio frame (%s)\n", av_err2str(ret));
+            fprintf(stderr, "Error decoding audio frame\n");
             return ret;
         }
         /* Some audio decoders decode only part of the packet, and have to be
@@ -145,22 +113,16 @@ static int decode_packet(int *got_frame, int cached)
         }
     }
 
-    /* If we use the new API with reference counting, we own the data and need
-     * to de-reference it when we don't use it anymore */
-    if (*got_frame && api_mode == API_MODE_NEW_API_REF_COUNT)
-        av_frame_unref(frame);
-
     return decoded;
 }
 
 static int open_codec_context(int *stream_idx,
                               AVFormatContext *fmt_ctx, enum AVMediaType type)
 {
-    int ret, stream_index;
+    int ret;
     AVStream *st;
     AVCodecContext *dec_ctx = NULL;
     AVCodec *dec = NULL;
-    AVDictionary *opts = NULL;
 
     ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
     if (ret < 0) {
@@ -168,8 +130,8 @@ static int open_codec_context(int *stream_idx,
                 av_get_media_type_string(type), src_filename);
         return ret;
     } else {
-        stream_index = ret;
-        st = fmt_ctx->streams[stream_index];
+        *stream_idx = ret;
+        st = fmt_ctx->streams[*stream_idx];
 
         /* find decoder for the stream */
         dec_ctx = st->codec;
@@ -177,18 +139,14 @@ static int open_codec_context(int *stream_idx,
         if (!dec) {
             fprintf(stderr, "Failed to find %s codec\n",
                     av_get_media_type_string(type));
-            return AVERROR(EINVAL);
+            return ret;
         }
 
-        /* Init the decoders, with or without reference counting */
-        if (api_mode == API_MODE_NEW_API_REF_COUNT)
-            av_dict_set(&opts, "refcounted_frames", "1", 0);
-        if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) {
+        if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
             fprintf(stderr, "Failed to open %s codec\n",
                     av_get_media_type_string(type));
             return ret;
         }
-        *stream_idx = stream_index;
     }
 
     return 0;
@@ -227,31 +185,15 @@ int main (int argc, char **argv)
 {
     int ret = 0, got_frame;
 
-    if (argc != 4 && argc != 5) {
-        fprintf(stderr, "usage: %s [-refcount=<old|new_norefcount|new_refcount>] "
-                "input_file video_output_file audio_output_file\n"
+    if (argc != 4) {
+        fprintf(stderr, "usage: %s input_file video_output_file audio_output_file\n"
                 "API example program to show how to read frames from an input file.\n"
                 "This program reads frames from a file, decodes them, and writes decoded\n"
                 "video frames to a rawvideo file named video_output_file, and decoded\n"
-                "audio frames to a rawaudio file named audio_output_file.\n\n"
-                "If the -refcount option is specified, the program use the\n"
-                "reference counting frame system which allows keeping a copy of\n"
-                "the data for longer than one decode call. If unset, it's using\n"
-                "the classic old method.\n"
+                "audio frames to a rawaudio file named audio_output_file.\n"
                 "\n", argv[0]);
         exit(1);
     }
-    if (argc == 5) {
-        const char *mode = argv[1] + strlen("-refcount=");
-        if      (!strcmp(mode, "old"))            api_mode = API_MODE_OLD;
-        else if (!strcmp(mode, "new_norefcount")) api_mode = API_MODE_NEW_API_NO_REF_COUNT;
-        else if (!strcmp(mode, "new_refcount"))   api_mode = API_MODE_NEW_API_REF_COUNT;
-        else {
-            fprintf(stderr, "unknow mode '%s'\n", mode);
-            exit(1);
-        }
-        argv++;
-    }
     src_filename = argv[1];
     video_dst_filename = argv[2];
     audio_dst_filename = argv[3];
@@ -283,11 +225,9 @@ int main (int argc, char **argv)
         }
 
         /* allocate image where the decoded image will be put */
-        width = video_dec_ctx->width;
-        height = video_dec_ctx->height;
-        pix_fmt = video_dec_ctx->pix_fmt;
         ret = av_image_alloc(video_dst_data, video_dst_linesize,
-                             width, height, pix_fmt, 1);
+                             video_dec_ctx->width, video_dec_ctx->height,
+                             video_dec_ctx->pix_fmt, 1);
         if (ret < 0) {
             fprintf(stderr, "Could not allocate raw video buffer\n");
             goto end;
@@ -300,7 +240,7 @@ int main (int argc, char **argv)
         audio_dec_ctx = audio_stream->codec;
         audio_dst_file = fopen(audio_dst_filename, "wb");
         if (!audio_dst_file) {
-            fprintf(stderr, "Could not open destination file %s\n", audio_dst_filename);
+            fprintf(stderr, "Could not open destination file %s\n", video_dst_filename);
             ret = 1;
             goto end;
         }
@@ -315,12 +255,7 @@ int main (int argc, char **argv)
         goto end;
     }
 
-    /* When using the new API, you need to use the libavutil/frame.h API, while
-     * the classic frame management is available in libavcodec */
-    if (api_mode == API_MODE_OLD)
     frame = avcodec_alloc_frame();
-    else
-        frame = av_frame_alloc();
     if (!frame) {
         fprintf(stderr, "Could not allocate frame\n");
         ret = AVERROR(ENOMEM);
@@ -362,7 +297,7 @@ int main (int argc, char **argv)
     if (video_stream) {
         printf("Play the output video file with the command:\n"
                "ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
-               av_get_pix_fmt_name(pix_fmt), width, height,
+               av_get_pix_fmt_name(video_dec_ctx->pix_fmt), video_dec_ctx->width, video_dec_ctx->height,
                video_dst_filename);
     }
 
@@ -390,17 +325,16 @@ int main (int argc, char **argv)
     }
 
 end:
+    if (video_dec_ctx)
         avcodec_close(video_dec_ctx);
+    if (audio_dec_ctx)
         avcodec_close(audio_dec_ctx);
     avformat_close_input(&fmt_ctx);
     if (video_dst_file)
         fclose(video_dst_file);
     if (audio_dst_file)
         fclose(audio_dst_file);
-    if (api_mode == API_MODE_OLD)
-        avcodec_free_frame(&frame);
-    else
-        av_frame_free(&frame);
+    av_free(frame);
     av_free(video_dst_data[0]);
 
     return ret < 0;

doc/examples/extract_mvs.c

@@ -1,185 +0,0 @@
-/*
- * Copyright (c) 2012 Stefano Sabatini
- * Copyright (c) 2014 Clément Bœsch
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
- * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- * THE SOFTWARE.
- */
-
-#include <libavutil/motion_vector.h>
-#include <libavformat/avformat.h>
-
-static AVFormatContext *fmt_ctx = NULL;
-static AVCodecContext *video_dec_ctx = NULL;
-static AVStream *video_stream = NULL;
-static const char *src_filename = NULL;
-
-static int video_stream_idx = -1;
-static AVFrame *frame = NULL;
-static AVPacket pkt;
-static int video_frame_count = 0;
-
-static int decode_packet(int *got_frame, int cached)
-{
-    int decoded = pkt.size;
-
-    *got_frame = 0;
-
-    if (pkt.stream_index == video_stream_idx) {
-        int ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
-        if (ret < 0) {
-            fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
-            return ret;
-        }
-
-        if (*got_frame) {
-            int i;
-            AVFrameSideData *sd;
-
-            video_frame_count++;
-            sd = av_frame_get_side_data(frame, AV_FRAME_DATA_MOTION_VECTORS);
-            if (sd) {
-                const AVMotionVector *mvs = (const AVMotionVector *)sd->data;
-                for (i = 0; i < sd->size / sizeof(*mvs); i++) {
-                    const AVMotionVector *mv = &mvs[i];
-                    printf("%d,%2d,%2d,%2d,%4d,%4d,%4d,%4d,0x%"PRIx64"\n",
-                           video_frame_count, mv->source,
-                           mv->w, mv->h, mv->src_x, mv->src_y,
-                           mv->dst_x, mv->dst_y, mv->flags);
-                }
-            }
-        }
-    }
-
-    return decoded;
-}
-
-static int open_codec_context(int *stream_idx,
-                              AVFormatContext *fmt_ctx, enum AVMediaType type)
-{
-    int ret;
-    AVStream *st;
-    AVCodecContext *dec_ctx = NULL;
-    AVCodec *dec = NULL;
-    AVDictionary *opts = NULL;
-
-    ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
-    if (ret < 0) {
-        fprintf(stderr, "Could not find %s stream in input file '%s'\n",
-                av_get_media_type_string(type), src_filename);
-        return ret;
-    } else {
-        *stream_idx = ret;
-        st = fmt_ctx->streams[*stream_idx];
-
-        /* find decoder for the stream */
-        dec_ctx = st->codec;
-        dec = avcodec_find_decoder(dec_ctx->codec_id);
-        if (!dec) {
-            fprintf(stderr, "Failed to find %s codec\n",
-                    av_get_media_type_string(type));
-            return AVERROR(EINVAL);
-        }
-
-        /* Init the video decoder */
-        av_dict_set(&opts, "flags2", "+export_mvs", 0);
-        if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) {
-            fprintf(stderr, "Failed to open %s codec\n",
-                    av_get_media_type_string(type));
-            return ret;
-        }
-    }
-
-    return 0;
-}
-
-int main(int argc, char **argv)
-{
-    int ret = 0, got_frame;
-
-    if (argc != 2) {
-        fprintf(stderr, "Usage: %s <video>\n", argv[0]);
-        exit(1);
-    }
-    src_filename = argv[1];
-
-    av_register_all();
-
-    if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
-        fprintf(stderr, "Could not open source file %s\n", src_filename);
-        exit(1);
-    }
-
-    if (avformat_find_stream_info(fmt_ctx, NULL) < 0) {
-        fprintf(stderr, "Could not find stream information\n");
-        exit(1);
-    }
-
-    if (open_codec_context(&video_stream_idx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
-        video_stream = fmt_ctx->streams[video_stream_idx];
-        video_dec_ctx = video_stream->codec;
-    }
-
-    av_dump_format(fmt_ctx, 0, src_filename, 0);
-
-    if (!video_stream) {
-        fprintf(stderr, "Could not find video stream in the input, aborting\n");
-        ret = 1;
-        goto end;
-    }
-
-    frame = av_frame_alloc();
-    if (!frame) {
-        fprintf(stderr, "Could not allocate frame\n");
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-
-    printf("framenum,source,blockw,blockh,srcx,srcy,dstx,dsty,flags\n");
-
-    /* initialize packet, set data to NULL, let the demuxer fill it */
-    av_init_packet(&pkt);
-    pkt.data = NULL;
-    pkt.size = 0;
-
-    /* read frames from the file */
-    while (av_read_frame(fmt_ctx, &pkt) >= 0) {
-        AVPacket orig_pkt = pkt;
-        do {
-            ret = decode_packet(&got_frame, 0);
-            if (ret < 0)
-                break;
-            pkt.data += ret;
-            pkt.size -= ret;
-        } while (pkt.size > 0);
-        av_free_packet(&orig_pkt);
-    }
-
-    /* flush cached frames */
-    pkt.data = NULL;
-    pkt.size = 0;
-    do {
-        decode_packet(&got_frame, 1);
-    } while (got_frame);
-
-end:
-    avcodec_close(video_dec_ctx);
-    avformat_close_input(&fmt_ctx);
-    av_frame_free(&frame);
-    return ret < 0;
-}

doc/examples/filter_audio.c

@@ -1,365 +0,0 @@
-/*
- * copyright (c) 2013 Andrew Kelley
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file
- * libavfilter API usage example.
- *
- * @example filter_audio.c
- * This example will generate a sine wave audio,
- * pass it through a simple filter chain, and then compute the MD5 checksum of
- * the output data.
- *
- * The filter chain it uses is:
- * (input) -> abuffer -> volume -> aformat -> abuffersink -> (output)
- *
- * abuffer: This provides the endpoint where you can feed the decoded samples.
- * volume: In this example we hardcode it to 0.90.
- * aformat: This converts the samples to the samplefreq, channel layout,
- *          and sample format required by the audio device.
- * abuffersink: This provides the endpoint where you can read the samples after
- *              they have passed through the filter chain.
- */
-
-#include <inttypes.h>
-#include <math.h>
-#include <stdio.h>
-#include <stdlib.h>
-
-#include "libavutil/channel_layout.h"
-#include "libavutil/md5.h"
-#include "libavutil/mem.h"
-#include "libavutil/opt.h"
-#include "libavutil/samplefmt.h"
-
-#include "libavfilter/avfilter.h"
-#include "libavfilter/buffersink.h"
-#include "libavfilter/buffersrc.h"
-
-#define INPUT_SAMPLERATE     48000
-#define INPUT_FORMAT         AV_SAMPLE_FMT_FLTP
-#define INPUT_CHANNEL_LAYOUT AV_CH_LAYOUT_5POINT0
-
-#define VOLUME_VAL 0.90
-
-static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
-                             AVFilterContext **sink)
-{
-    AVFilterGraph *filter_graph;
-    AVFilterContext *abuffer_ctx;
-    AVFilter        *abuffer;
-    AVFilterContext *volume_ctx;
-    AVFilter        *volume;
-    AVFilterContext *aformat_ctx;
-    AVFilter        *aformat;
-    AVFilterContext *abuffersink_ctx;
-    AVFilter        *abuffersink;
-
-    AVDictionary *options_dict = NULL;
-    uint8_t options_str[1024];
-    uint8_t ch_layout[64];
-
-    int err;
-
-    /* Create a new filtergraph, which will contain all the filters. */
-    filter_graph = avfilter_graph_alloc();
-    if (!filter_graph) {
-        fprintf(stderr, "Unable to create filter graph.\n");
-        return AVERROR(ENOMEM);
-    }
-
-    /* Create the abuffer filter;
-     * it will be used for feeding the data into the graph. */
-    abuffer = avfilter_get_by_name("abuffer");
-    if (!abuffer) {
-        fprintf(stderr, "Could not find the abuffer filter.\n");
-        return AVERROR_FILTER_NOT_FOUND;
-    }
-
-    abuffer_ctx = avfilter_graph_alloc_filter(filter_graph, abuffer, "src");
-    if (!abuffer_ctx) {
-        fprintf(stderr, "Could not allocate the abuffer instance.\n");
-        return AVERROR(ENOMEM);
-    }
-
-    /* Set the filter options through the AVOptions API. */
-    av_get_channel_layout_string(ch_layout, sizeof(ch_layout), 0, INPUT_CHANNEL_LAYOUT);
-    av_opt_set    (abuffer_ctx, "channel_layout", ch_layout,                            AV_OPT_SEARCH_CHILDREN);
-    av_opt_set    (abuffer_ctx, "sample_fmt",     av_get_sample_fmt_name(INPUT_FORMAT), AV_OPT_SEARCH_CHILDREN);
-    av_opt_set_q  (abuffer_ctx, "time_base",      (AVRational){ 1, INPUT_SAMPLERATE },  AV_OPT_SEARCH_CHILDREN);
-    av_opt_set_int(abuffer_ctx, "sample_rate",    INPUT_SAMPLERATE,                     AV_OPT_SEARCH_CHILDREN);
-
-    /* Now initialize the filter; we pass NULL options, since we have already
-     * set all the options above. */
-    err = avfilter_init_str(abuffer_ctx, NULL);
-    if (err < 0) {
-        fprintf(stderr, "Could not initialize the abuffer filter.\n");
-        return err;
-    }
-
-    /* Create volume filter. */
-    volume = avfilter_get_by_name("volume");
-    if (!volume) {
-        fprintf(stderr, "Could not find the volume filter.\n");
-        return AVERROR_FILTER_NOT_FOUND;
-    }
-
-    volume_ctx = avfilter_graph_alloc_filter(filter_graph, volume, "volume");
-    if (!volume_ctx) {
-        fprintf(stderr, "Could not allocate the volume instance.\n");
-        return AVERROR(ENOMEM);
-    }
-
-    /* A different way of passing the options is as key/value pairs in a
-     * dictionary. */
-    av_dict_set(&options_dict, "volume", AV_STRINGIFY(VOLUME_VAL), 0);
-    err = avfilter_init_dict(volume_ctx, &options_dict);
-    av_dict_free(&options_dict);
-    if (err < 0) {
-        fprintf(stderr, "Could not initialize the volume filter.\n");
-        return err;
-    }
-
-    /* Create the aformat filter;
-     * it ensures that the output is of the format we want. */
-    aformat = avfilter_get_by_name("aformat");
-    if (!aformat) {
-        fprintf(stderr, "Could not find the aformat filter.\n");
-        return AVERROR_FILTER_NOT_FOUND;
-    }
-
-    aformat_ctx = avfilter_graph_alloc_filter(filter_graph, aformat, "aformat");
-    if (!aformat_ctx) {
-        fprintf(stderr, "Could not allocate the aformat instance.\n");
-        return AVERROR(ENOMEM);
-    }
-
-    /* A third way of passing the options is in a string of the form
-     * key1=value1:key2=value2.... */
-    snprintf(options_str, sizeof(options_str),
-             "sample_fmts=%s:sample_rates=%d:channel_layouts=0x%"PRIx64,
-             av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), 44100,
-             (uint64_t)AV_CH_LAYOUT_STEREO);
-    err = avfilter_init_str(aformat_ctx, options_str);
-    if (err < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Could not initialize the aformat filter.\n");
-        return err;
-    }
-
-    /* Finally create the abuffersink filter;
-     * it will be used to get the filtered data out of the graph. */
-    abuffersink = avfilter_get_by_name("abuffersink");
-    if (!abuffersink) {
-        fprintf(stderr, "Could not find the abuffersink filter.\n");
-        return AVERROR_FILTER_NOT_FOUND;
-    }
-
-    abuffersink_ctx = avfilter_graph_alloc_filter(filter_graph, abuffersink, "sink");
-    if (!abuffersink_ctx) {
-        fprintf(stderr, "Could not allocate the abuffersink instance.\n");
-        return AVERROR(ENOMEM);
-    }
-
-    /* This filter takes no options. */
-    err = avfilter_init_str(abuffersink_ctx, NULL);
-    if (err < 0) {
-        fprintf(stderr, "Could not initialize the abuffersink instance.\n");
-        return err;
-    }
-
-    /* Connect the filters;
-     * in this simple case the filters just form a linear chain. */
-    err = avfilter_link(abuffer_ctx, 0, volume_ctx, 0);
-    if (err >= 0)
-        err = avfilter_link(volume_ctx, 0, aformat_ctx, 0);
-    if (err >= 0)
-        err = avfilter_link(aformat_ctx, 0, abuffersink_ctx, 0);
-    if (err < 0) {
-        fprintf(stderr, "Error connecting filters\n");
-        return err;
-    }
-
-    /* Configure the graph. */
-    err = avfilter_graph_config(filter_graph, NULL);
-    if (err < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Error configuring the filter graph\n");
-        return err;
-    }
-
-    *graph = filter_graph;
-    *src   = abuffer_ctx;
-    *sink  = abuffersink_ctx;
-
-    return 0;
-}
-
-/* Do something useful with the filtered data: this simple
- * example just prints the MD5 checksum of each plane to stdout. */
-static int process_output(struct AVMD5 *md5, AVFrame *frame)
-{
-    int planar     = av_sample_fmt_is_planar(frame->format);
-    int channels   = av_get_channel_layout_nb_channels(frame->channel_layout);
-    int planes     = planar ? channels : 1;
-    int bps        = av_get_bytes_per_sample(frame->format);
-    int plane_size = bps * frame->nb_samples * (planar ? 1 : channels);
-    int i, j;
-
-    for (i = 0; i < planes; i++) {
-        uint8_t checksum[16];
-
-        av_md5_init(md5);
-        av_md5_sum(checksum, frame->extended_data[i], plane_size);
-
-        fprintf(stdout, "plane %d: 0x", i);
-        for (j = 0; j < sizeof(checksum); j++)
-            fprintf(stdout, "%02X", checksum[j]);
-        fprintf(stdout, "\n");
-    }
-    fprintf(stdout, "\n");
-
-    return 0;
-}
-
-/* Construct a frame of audio data to be filtered;
- * this simple example just synthesizes a sine wave. */
-static int get_input(AVFrame *frame, int frame_num)
-{
-    int err, i, j;
-
-#define FRAME_SIZE 1024
-
-    /* Set up the frame properties and allocate the buffer for the data. */
-    frame->sample_rate    = INPUT_SAMPLERATE;
-    frame->format         = INPUT_FORMAT;
-    frame->channel_layout = INPUT_CHANNEL_LAYOUT;
-    frame->nb_samples     = FRAME_SIZE;
-    frame->pts            = frame_num * FRAME_SIZE;
-
-    err = av_frame_get_buffer(frame, 0);
-    if (err < 0)
-        return err;
-
-    /* Fill the data for each channel. */
-    for (i = 0; i < 5; i++) {
-        float *data = (float*)frame->extended_data[i];
-
-        for (j = 0; j < frame->nb_samples; j++)
-            data[j] = sin(2 * M_PI * (frame_num + j) * (i + 1) / FRAME_SIZE);
-    }
-
-    return 0;
-}
-
-int main(int argc, char *argv[])
-{
-    struct AVMD5 *md5;
-    AVFilterGraph *graph;
-    AVFilterContext *src, *sink;
-    AVFrame *frame;
-    uint8_t errstr[1024];
-    float duration;
-    int err, nb_frames, i;
-
-    if (argc < 2) {
-        fprintf(stderr, "Usage: %s <duration>\n", argv[0]);
-        return 1;
-    }
-
-    duration  = atof(argv[1]);
-    nb_frames = duration * INPUT_SAMPLERATE / FRAME_SIZE;
-    if (nb_frames <= 0) {
-        fprintf(stderr, "Invalid duration: %s\n", argv[1]);
-        return 1;
-    }
-
-    avfilter_register_all();
-
-    /* Allocate the frame we will be using to store the data. */
-    frame  = av_frame_alloc();
-    if (!frame) {
-        fprintf(stderr, "Error allocating the frame\n");
-        return 1;
-    }
-
-    md5 = av_md5_alloc();
-    if (!md5) {
-        fprintf(stderr, "Error allocating the MD5 context\n");
-        return 1;
-    }
-
-    /* Set up the filtergraph. */
-    err = init_filter_graph(&graph, &src, &sink);
-    if (err < 0) {
-        fprintf(stderr, "Unable to init filter graph:");
-        goto fail;
-    }
-
-    /* the main filtering loop */
-    for (i = 0; i < nb_frames; i++) {
-        /* get an input frame to be filtered */
-        err = get_input(frame, i);
-        if (err < 0) {
-            fprintf(stderr, "Error generating input frame:");
-            goto fail;
-        }
-
-        /* Send the frame to the input of the filtergraph. */
-        err = av_buffersrc_add_frame(src, frame);
-        if (err < 0) {
-            av_frame_unref(frame);
-            fprintf(stderr, "Error submitting the frame to the filtergraph:");
-            goto fail;
-        }
-
-        /* Get all the filtered output that is available. */
-        while ((err = av_buffersink_get_frame(sink, frame)) >= 0) {
-            /* now do something with our filtered frame */
-            err = process_output(md5, frame);
-            if (err < 0) {
-                fprintf(stderr, "Error processing the filtered frame:");
-                goto fail;
-            }
-            av_frame_unref(frame);
-        }
-
-        if (err == AVERROR(EAGAIN)) {
-            /* Need to feed more frames in. */
-            continue;
-        } else if (err == AVERROR_EOF) {
-            /* Nothing more to do, finish. */
-            break;
-        } else if (err < 0) {
-            /* An error occurred. */
-            fprintf(stderr, "Error filtering the data:");
-            goto fail;
-        }
-    }
-
-    avfilter_graph_free(&graph);
-    av_frame_free(&frame);
-    av_freep(&md5);
-
-    return 0;
-
-fail:
-    av_strerror(err, errstr, sizeof(errstr));
-    fprintf(stderr, "%s\n", errstr);
-    return 1;
-}

doc/examples/filtering_audio.c

@@ -25,7 +25,7 @@
 /**
  * @file
  * API example for audio decoding and filtering
- * @example filtering_audio.c
+ * @example doc/examples/filtering_audio.c
  */
 
 #include <unistd.h>
@@ -85,7 +85,7 @@ static int open_input_file(const char *filename)
 static int init_filters(const char *filters_descr)
 {
     char args[512];
-    int ret = 0;
+    int ret;
     AVFilter *abuffersrc  = avfilter_get_by_name("abuffer");
     AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
     AVFilterInOut *outputs = avfilter_inout_alloc();
@@ -97,10 +97,6 @@ static int init_filters(const char *filters_descr)
     AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;
 
     filter_graph = avfilter_graph_alloc();
-    if (!outputs || !inputs || !filter_graph) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
 
     /* buffer audio source: the decoded frames from the decoder will be inserted here. */
     if (!dec_ctx->channel_layout)
@@ -113,7 +109,7 @@ static int init_filters(const char *filters_descr)
                                        args, NULL, filter_graph);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
-        goto end;
+        return ret;
     }
 
     /* buffer audio sink: to terminate the filter chain. */
@@ -121,52 +117,36 @@ static int init_filters(const char *filters_descr)
                                        NULL, NULL, filter_graph);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
-        goto end;
+        return ret;
     }
 
     ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1,
                               AV_OPT_SEARCH_CHILDREN);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
-        goto end;
+        return ret;
     }
 
     ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1,
                               AV_OPT_SEARCH_CHILDREN);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
-        goto end;
+        return ret;
     }
 
     ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -1,
                               AV_OPT_SEARCH_CHILDREN);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
-        goto end;
+        return ret;
     }
 
-    /*
-     * Set the endpoints for the filter graph. The filter_graph will
-     * be linked to the graph described by filters_descr.
-     */
-
-    /*
-     * The buffer source output must be connected to the input pad of
-     * the first filter described by filters_descr; since the first
-     * filter input label is not specified, it is set to "in" by
-     * default.
-     */
+    /* Endpoints for the filter graph. */
     outputs->name       = av_strdup("in");
     outputs->filter_ctx = buffersrc_ctx;
     outputs->pad_idx    = 0;
     outputs->next       = NULL;
 
-    /*
-     * The buffer sink input must be connected to the output pad of
-     * the last filter described by filters_descr; since the last
-     * filter output label is not specified, it is set to "out" by
-     * default.
-     */
     inputs->name       = av_strdup("out");
     inputs->filter_ctx = buffersink_ctx;
     inputs->pad_idx    = 0;
@@ -174,10 +154,10 @@ static int init_filters(const char *filters_descr)
 
     if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
                                     &inputs, &outputs, NULL)) < 0)
-        goto end;
+        return ret;
 
     if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
-        goto end;
+        return ret;
 
     /* Print summary of the sink buffer
      * Note: args buffer is reused to store channel layout string */
@@ -188,11 +168,7 @@ static int init_filters(const char *filters_descr)
            (char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
            args);
 
-end:
-    avfilter_inout_free(&inputs);
-    avfilter_inout_free(&outputs);
-
-    return ret;
+    return 0;
 }
 
 static void print_frame(const AVFrame *frame)
@@ -212,7 +188,7 @@ static void print_frame(const AVFrame *frame)
 int main(int argc, char **argv)
 {
     int ret;
-    AVPacket packet0, packet;
+    AVPacket packet;
     AVFrame *frame = av_frame_alloc();
     AVFrame *filt_frame = av_frame_alloc();
     int got_frame;
@@ -226,6 +202,7 @@ int main(int argc, char **argv)
         exit(1);
     }
 
+    avcodec_register_all();
     av_register_all();
     avfilter_register_all();
 
@@ -235,24 +212,18 @@ int main(int argc, char **argv)
         goto end;
 
     /* read all packets */
-    packet0.data = NULL;
-    packet.data = NULL;
     while (1) {
-        if (!packet0.data) {
         if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
             break;
-            packet0 = packet;
-        }
 
         if (packet.stream_index == audio_stream_index) {
+            avcodec_get_frame_defaults(frame);
             got_frame = 0;
             ret = avcodec_decode_audio4(dec_ctx, frame, &got_frame, &packet);
             if (ret < 0) {
                 av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
                 continue;
             }
-            packet.size -= ret;
-            packet.data += ret;
 
             if (got_frame) {
                 /* push the audio data from decoded frame into the filtergraph */
@@ -272,23 +243,21 @@ int main(int argc, char **argv)
                     av_frame_unref(filt_frame);
                 }
             }
-
-            if (packet.size <= 0)
-                av_free_packet(&packet0);
-        } else {
-            /* discard non-wanted packets */
-            av_free_packet(&packet0);
         }
+        av_free_packet(&packet);
     }
 end:
     avfilter_graph_free(&filter_graph);
+    if (dec_ctx)
         avcodec_close(dec_ctx);
     avformat_close_input(&fmt_ctx);
     av_frame_free(&frame);
     av_frame_free(&filt_frame);
 
     if (ret < 0 && ret != AVERROR_EOF) {
-        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
+        char buf[1024];
+        av_strerror(ret, buf, sizeof(buf));
+        fprintf(stderr, "Error occurred: %s\n", buf);
         exit(1);
     }
 

doc/examples/filtering_video.c

@@ -24,7 +24,7 @@
 /**
  * @file
  * API example for decoding and filtering
- * @example filtering_video.c
+ * @example doc/examples/filtering_video.c
  */
 
 #define _XOPEN_SOURCE 600 /* for usleep */
@@ -36,7 +36,6 @@
 #include <libavfilter/avcodec.h>
 #include <libavfilter/buffersink.h>
 #include <libavfilter/buffersrc.h>
-#include <libavutil/opt.h>
 
 const char *filter_descr = "scale=78:24";
 
@@ -71,7 +70,6 @@ static int open_input_file(const char *filename)
     }
     video_stream_index = ret;
     dec_ctx = fmt_ctx->streams[video_stream_index]->codec;
-    av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
 
     /* init the video decoder */
     if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
@@ -85,71 +83,47 @@ static int open_input_file(const char *filename)
 static int init_filters(const char *filters_descr)
 {
     char args[512];
-    int ret = 0;
+    int ret;
     AVFilter *buffersrc  = avfilter_get_by_name("buffer");
     AVFilter *buffersink = avfilter_get_by_name("buffersink");
     AVFilterInOut *outputs = avfilter_inout_alloc();
     AVFilterInOut *inputs  = avfilter_inout_alloc();
-    AVRational time_base = fmt_ctx->streams[video_stream_index]->time_base;
     enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE };
+    AVBufferSinkParams *buffersink_params;
 
     filter_graph = avfilter_graph_alloc();
-    if (!outputs || !inputs || !filter_graph) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
 
     /* buffer video source: the decoded frames from the decoder will be inserted here. */
     snprintf(args, sizeof(args),
             "video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
             dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
-            time_base.num, time_base.den,
+            dec_ctx->time_base.num, dec_ctx->time_base.den,
             dec_ctx->sample_aspect_ratio.num, dec_ctx->sample_aspect_ratio.den);
 
     ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
                                        args, NULL, filter_graph);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
-        goto end;
+        return ret;
     }
 
     /* buffer video sink: to terminate the filter chain. */
+    buffersink_params = av_buffersink_params_alloc();
+    buffersink_params->pixel_fmts = pix_fmts;
     ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
-                                       NULL, NULL, filter_graph);
+                                       NULL, buffersink_params, filter_graph);
+    av_free(buffersink_params);
     if (ret < 0) {
         av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
-        goto end;
+        return ret;
     }
 
-    ret = av_opt_set_int_list(buffersink_ctx, "pix_fmts", pix_fmts,
-                              AV_PIX_FMT_NONE, AV_OPT_SEARCH_CHILDREN);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n");
-        goto end;
-    }
-
-    /*
-     * Set the endpoints for the filter graph. The filter_graph will
-     * be linked to the graph described by filters_descr.
-     */
-
-    /*
-     * The buffer source output must be connected to the input pad of
-     * the first filter described by filters_descr; since the first
-     * filter input label is not specified, it is set to "in" by
-     * default.
-     */
+    /* Endpoints for the filter graph. */
     outputs->name       = av_strdup("in");
     outputs->filter_ctx = buffersrc_ctx;
     outputs->pad_idx    = 0;
     outputs->next       = NULL;
 
-    /*
-     * The buffer sink input must be connected to the output pad of
-     * the last filter described by filters_descr; since the last
-     * filter output label is not specified, it is set to "out" by
-     * default.
-     */
     inputs->name       = av_strdup("out");
     inputs->filter_ctx = buffersink_ctx;
     inputs->pad_idx    = 0;
@@ -157,16 +131,11 @@ static int init_filters(const char *filters_descr)
 
     if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
                                     &inputs, &outputs, NULL)) < 0)
-        goto end;
+        return ret;
 
     if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
-        goto end;
-
-end:
-    avfilter_inout_free(&inputs);
-    avfilter_inout_free(&outputs);
-
         return ret;
+    return 0;
 }
 
 static void display_frame(const AVFrame *frame, AVRational time_base)
@@ -217,6 +186,7 @@ int main(int argc, char **argv)
         exit(1);
     }
 
+    avcodec_register_all();
     av_register_all();
     avfilter_register_all();
 
@@ -231,6 +201,7 @@ int main(int argc, char **argv)
             break;
 
         if (packet.stream_index == video_stream_index) {
+            avcodec_get_frame_defaults(frame);
             got_frame = 0;
             ret = avcodec_decode_video2(dec_ctx, frame, &got_frame, &packet);
             if (ret < 0) {
@@ -257,20 +228,22 @@ int main(int argc, char **argv)
                     display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base);
                     av_frame_unref(filt_frame);
                 }
-                av_frame_unref(frame);
             }
         }
         av_free_packet(&packet);
     }
 end:
     avfilter_graph_free(&filter_graph);
+    if (dec_ctx)
         avcodec_close(dec_ctx);
     avformat_close_input(&fmt_ctx);
     av_frame_free(&frame);
     av_frame_free(&filt_frame);
 
     if (ret < 0 && ret != AVERROR_EOF) {
-        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
+        char buf[1024];
+        av_strerror(ret, buf, sizeof(buf));
+        fprintf(stderr, "Error occurred: %s\n", buf);
         exit(1);
     }
 

doc/examples/metadata.c

@@ -23,7 +23,7 @@
 /**
  * @file
  * Shows how the metadata API can be used in application programs.
- * @example metadata.c
+ * @example doc/examples/metadata.c
  */
 
 #include <stdio.h>

doc/examples/muxing.c

@@ -24,9 +24,9 @@
  * @file
  * libavformat API example.
  *
- * Output a media file in any supported libavformat format. The default
- * codecs are used.
- * @example muxing.c
+ * Output a media file in any supported libavformat format.
+ * The default codecs are used.
+ * @example doc/examples/muxing.c
  */
 
 #include <stdlib.h>
@@ -34,67 +34,26 @@
 #include <string.h>
 #include <math.h>
 
-#include <libavutil/avassert.h>
-#include <libavutil/channel_layout.h>
 #include <libavutil/opt.h>
 #include <libavutil/mathematics.h>
-#include <libavutil/timestamp.h>
 #include <libavformat/avformat.h>
 #include <libswscale/swscale.h>
 #include <libswresample/swresample.h>
 
-#define STREAM_DURATION   10.0
+/* 5 seconds stream duration */
+#define STREAM_DURATION   200.0
 #define STREAM_FRAME_RATE 25 /* 25 images/s */
+#define STREAM_NB_FRAMES  ((int)(STREAM_DURATION * STREAM_FRAME_RATE))
 #define STREAM_PIX_FMT    AV_PIX_FMT_YUV420P /* default pix_fmt */
 
-#define SCALE_FLAGS SWS_BICUBIC
-
-// a wrapper around a single output AVStream
-typedef struct OutputStream {
-    AVStream *st;
-
-    /* pts of the next frame that will be generated */
-    int64_t next_pts;
-    int samples_count;
-
-    AVFrame *frame;
-    AVFrame *tmp_frame;
-
-    float t, tincr, tincr2;
-
-    struct SwsContext *sws_ctx;
-    struct SwrContext *swr_ctx;
-} OutputStream;
-
-static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
-{
-    AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
-
-    printf("pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
-           av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base),
-           av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base),
-           av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base),
-           pkt->stream_index);
-}
-
-static int write_frame(AVFormatContext *fmt_ctx, const AVRational *time_base, AVStream *st, AVPacket *pkt)
-{
-    /* rescale output packet timestamp values from codec to stream timebase */
-    av_packet_rescale_ts(pkt, *time_base, st->time_base);
-    pkt->stream_index = st->index;
-
-    /* Write the compressed frame to the media file. */
-    log_packet(fmt_ctx, pkt);
-    return av_interleaved_write_frame(fmt_ctx, pkt);
-}
+static int sws_flags = SWS_BICUBIC;
 
 /* Add an output stream. */
-static void add_stream(OutputStream *ost, AVFormatContext *oc,
-                       AVCodec **codec,
+static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
                             enum AVCodecID codec_id)
 {
     AVCodecContext *c;
-    int i;
+    AVStream *st;
 
     /* find the encoder */
     *codec = avcodec_find_encoder(codec_id);
@@ -104,38 +63,20 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
         exit(1);
     }
 
-    ost->st = avformat_new_stream(oc, *codec);
-    if (!ost->st) {
+    st = avformat_new_stream(oc, *codec);
+    if (!st) {
         fprintf(stderr, "Could not allocate stream\n");
         exit(1);
     }
-    ost->st->id = oc->nb_streams-1;
-    c = ost->st->codec;
+    st->id = oc->nb_streams-1;
+    c = st->codec;
 
     switch ((*codec)->type) {
     case AVMEDIA_TYPE_AUDIO:
-        c->sample_fmt  = (*codec)->sample_fmts ?
-            (*codec)->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
+        c->sample_fmt  = AV_SAMPLE_FMT_FLTP;
         c->bit_rate    = 64000;
         c->sample_rate = 44100;
-        if ((*codec)->supported_samplerates) {
-            c->sample_rate = (*codec)->supported_samplerates[0];
-            for (i = 0; (*codec)->supported_samplerates[i]; i++) {
-                if ((*codec)->supported_samplerates[i] == 44100)
-                    c->sample_rate = 44100;
-            }
-        }
-        c->channels        = av_get_channel_layout_nb_channels(c->channel_layout);
-        c->channel_layout = AV_CH_LAYOUT_STEREO;
-        if ((*codec)->channel_layouts) {
-            c->channel_layout = (*codec)->channel_layouts[0];
-            for (i = 0; (*codec)->channel_layouts[i]; i++) {
-                if ((*codec)->channel_layouts[i] == AV_CH_LAYOUT_STEREO)
-                    c->channel_layout = AV_CH_LAYOUT_STEREO;
-            }
-        }
-        c->channels        = av_get_channel_layout_nb_channels(c->channel_layout);
-        ost->st->time_base = (AVRational){ 1, c->sample_rate };
+        c->channels    = 2;
         break;
 
     case AVMEDIA_TYPE_VIDEO:
@@ -149,9 +90,8 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
          * of which frame timestamps are represented. For fixed-fps content,
          * timebase should be 1/framerate and timestamp increments should be
          * identical to 1. */
-        ost->st->time_base = (AVRational){ 1, STREAM_FRAME_RATE };
-        c->time_base       = ost->st->time_base;
-
+        c->time_base.den = STREAM_FRAME_RATE;
+        c->time_base.num = 1;
         c->gop_size      = 12; /* emit one intra frame every twelve frames at most */
         c->pix_fmt       = STREAM_PIX_FMT;
         if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
@@ -173,262 +113,237 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
     /* Some formats want stream headers to be separate. */
     if (oc->oformat->flags & AVFMT_GLOBALHEADER)
         c->flags |= CODEC_FLAG_GLOBAL_HEADER;
+
+    return st;
 }
 
 /**************************************************************/
 /* audio output */
 
-static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt,
-                                  uint64_t channel_layout,
-                                  int sample_rate, int nb_samples)
-{
-    AVFrame *frame = av_frame_alloc();
-    int ret;
+static float t, tincr, tincr2;
 
-    if (!frame) {
-        fprintf(stderr, "Error allocating an audio frame\n");
-        exit(1);
-    }
+static uint8_t **src_samples_data;
+static int       src_samples_linesize;
+static int       src_nb_samples;
 
-    frame->format = sample_fmt;
-    frame->channel_layout = channel_layout;
-    frame->sample_rate = sample_rate;
-    frame->nb_samples = nb_samples;
+static int max_dst_nb_samples;
+uint8_t **dst_samples_data;
+int       dst_samples_linesize;
+int       dst_samples_size;
 
-    if (nb_samples) {
-        ret = av_frame_get_buffer(frame, 0);
-        if (ret < 0) {
-            fprintf(stderr, "Error allocating an audio buffer\n");
-            exit(1);
-        }
-    }
+struct SwrContext *swr_ctx = NULL;
 
-    return frame;
-}
-
-static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
+static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
 {
     AVCodecContext *c;
-    int nb_samples;
     int ret;
-    AVDictionary *opt = NULL;
 
-    c = ost->st->codec;
+    c = st->codec;
 
     /* open it */
-    av_dict_copy(&opt, opt_arg, 0);
-    ret = avcodec_open2(c, codec, &opt);
-    av_dict_free(&opt);
+    ret = avcodec_open2(c, codec, NULL);
     if (ret < 0) {
         fprintf(stderr, "Could not open audio codec: %s\n", av_err2str(ret));
         exit(1);
     }
 
     /* init signal generator */
-    ost->t     = 0;
-    ost->tincr = 2 * M_PI * 110.0 / c->sample_rate;
+    t     = 0;
+    tincr = 2 * M_PI * 110.0 / c->sample_rate;
     /* increment frequency by 110 Hz per second */
-    ost->tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
+    tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
 
-    if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
-        nb_samples = 10000;
-    else
-        nb_samples = c->frame_size;
+    src_nb_samples = c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE ?
+        10000 : c->frame_size;
 
-    ost->frame     = alloc_audio_frame(c->sample_fmt, c->channel_layout,
-                                       c->sample_rate, nb_samples);
-    ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, c->channel_layout,
-                                       c->sample_rate, nb_samples);
+    ret = av_samples_alloc_array_and_samples(&src_samples_data, &src_samples_linesize, c->channels,
+                                             src_nb_samples, c->sample_fmt, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Could not allocate source samples\n");
+        exit(1);
+    }
 
     /* create resampler context */
-        ost->swr_ctx = swr_alloc();
-        if (!ost->swr_ctx) {
+    if (c->sample_fmt != AV_SAMPLE_FMT_S16) {
+        swr_ctx = swr_alloc();
+        if (!swr_ctx) {
             fprintf(stderr, "Could not allocate resampler context\n");
             exit(1);
         }
 
         /* set options */
-        av_opt_set_int       (ost->swr_ctx, "in_channel_count",   c->channels,       0);
-        av_opt_set_int       (ost->swr_ctx, "in_sample_rate",     c->sample_rate,    0);
-        av_opt_set_sample_fmt(ost->swr_ctx, "in_sample_fmt",      AV_SAMPLE_FMT_S16, 0);
-        av_opt_set_int       (ost->swr_ctx, "out_channel_count",  c->channels,       0);
-        av_opt_set_int       (ost->swr_ctx, "out_sample_rate",    c->sample_rate,    0);
-        av_opt_set_sample_fmt(ost->swr_ctx, "out_sample_fmt",     c->sample_fmt,     0);
+        av_opt_set_int       (swr_ctx, "in_channel_count",   c->channels,       0);
+        av_opt_set_int       (swr_ctx, "in_sample_rate",     c->sample_rate,    0);
+        av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt",      AV_SAMPLE_FMT_S16, 0);
+        av_opt_set_int       (swr_ctx, "out_channel_count",  c->channels,       0);
+        av_opt_set_int       (swr_ctx, "out_sample_rate",    c->sample_rate,    0);
+        av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt",     c->sample_fmt,     0);
 
         /* initialize the resampling context */
-        if ((ret = swr_init(ost->swr_ctx)) < 0) {
+        if ((ret = swr_init(swr_ctx)) < 0) {
             fprintf(stderr, "Failed to initialize the resampling context\n");
             exit(1);
         }
     }
 
+    /* compute the number of converted samples: buffering is avoided
+     * ensuring that the output buffer will contain at least all the
+     * converted input samples */
+    max_dst_nb_samples = src_nb_samples;
+    ret = av_samples_alloc_array_and_samples(&dst_samples_data, &dst_samples_linesize, c->channels,
+                                             max_dst_nb_samples, c->sample_fmt, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Could not allocate destination samples\n");
+        exit(1);
+    }
+    dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, max_dst_nb_samples,
+                                                  c->sample_fmt, 0);
+}
+
 /* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
  * 'nb_channels' channels. */
-static AVFrame *get_audio_frame(OutputStream *ost)
+static void get_audio_frame(int16_t *samples, int frame_size, int nb_channels)
 {
-    AVFrame *frame = ost->tmp_frame;
     int j, i, v;
-    int16_t *q = (int16_t*)frame->data[0];
+    int16_t *q;
 
-    /* check if we want to generate more frames */
-    if (av_compare_ts(ost->next_pts, ost->st->codec->time_base,
-                      STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
-        return NULL;
-
-    for (j = 0; j <frame->nb_samples; j++) {
-        v = (int)(sin(ost->t) * 10000);
-        for (i = 0; i < ost->st->codec->channels; i++)
+    q = samples;
+    for (j = 0; j < frame_size; j++) {
+        v = (int)(sin(t) * 10000);
+        for (i = 0; i < nb_channels; i++)
             *q++ = v;
-        ost->t     += ost->tincr;
-        ost->tincr += ost->tincr2;
+        t     += tincr;
+        tincr += tincr2;
+    }
 }
 
-    frame->pts = ost->next_pts;
-    ost->next_pts  += frame->nb_samples;
-
-    return frame;
-}
-
-/*
- * encode one audio frame and send it to the muxer
- * return 1 when encoding is finished, 0 otherwise
- */
-static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
+static void write_audio_frame(AVFormatContext *oc, AVStream *st)
 {
     AVCodecContext *c;
     AVPacket pkt = { 0 }; // data and size must be 0;
-    AVFrame *frame;
-    int ret;
-    int got_packet;
-    int dst_nb_samples;
+    AVFrame *frame = avcodec_alloc_frame();
+    int got_packet, ret, dst_nb_samples;
 
     av_init_packet(&pkt);
-    c = ost->st->codec;
+    c = st->codec;
 
-    frame = get_audio_frame(ost);
+    get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);
 
-    if (frame) {
     /* convert samples from native format to destination codec format, using the resampler */
+    if (swr_ctx) {
         /* compute destination number of samples */
-            dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples,
+        dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + src_nb_samples,
                                         c->sample_rate, c->sample_rate, AV_ROUND_UP);
-            av_assert0(dst_nb_samples == frame->nb_samples);
-
-        /* when we pass a frame to the encoder, it may keep a reference to it
-         * internally;
-         * make sure we do not overwrite it here
-         */
-        ret = av_frame_make_writable(ost->frame);
+        if (dst_nb_samples > max_dst_nb_samples) {
+            av_free(dst_samples_data[0]);
+            ret = av_samples_alloc(dst_samples_data, &dst_samples_linesize, c->channels,
+                                   dst_nb_samples, c->sample_fmt, 0);
             if (ret < 0)
                 exit(1);
+            max_dst_nb_samples = dst_nb_samples;
+            dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples,
+                                                          c->sample_fmt, 0);
+        }
 
         /* convert to destination format */
-            ret = swr_convert(ost->swr_ctx,
-                              ost->frame->data, dst_nb_samples,
-                              (const uint8_t **)frame->data, frame->nb_samples);
+        ret = swr_convert(swr_ctx,
+                          dst_samples_data, dst_nb_samples,
+                          (const uint8_t **)src_samples_data, src_nb_samples);
         if (ret < 0) {
             fprintf(stderr, "Error while converting\n");
             exit(1);
         }
-            frame = ost->frame;
-
-        frame->pts = av_rescale_q(ost->samples_count, (AVRational){1, c->sample_rate}, c->time_base);
-        ost->samples_count += dst_nb_samples;
+    } else {
+        dst_samples_data[0] = src_samples_data[0];
+        dst_nb_samples = src_nb_samples;
     }
 
+    frame->nb_samples = dst_nb_samples;
+    avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
+                             dst_samples_data[0], dst_samples_size, 0);
+
     ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
     if (ret < 0) {
         fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
         exit(1);
     }
 
-    if (got_packet) {
-        ret = write_frame(oc, &c->time_base, ost->st, &pkt);
-        if (ret < 0) {
+    if (!got_packet)
+        return;
+
+    pkt.stream_index = st->index;
+
+    /* Write the compressed frame to the media file. */
+    ret = av_interleaved_write_frame(oc, &pkt);
+    if (ret != 0) {
         fprintf(stderr, "Error while writing audio frame: %s\n",
                 av_err2str(ret));
         exit(1);
     }
+    avcodec_free_frame(&frame);
 }
 
-    return (frame || got_packet) ? 0 : 1;
+static void close_audio(AVFormatContext *oc, AVStream *st)
+{
+    avcodec_close(st->codec);
+    av_free(src_samples_data[0]);
+    av_free(dst_samples_data[0]);
 }
 
 /**************************************************************/
 /* video output */
 
-static AVFrame *alloc_picture(enum AVPixelFormat pix_fmt, int width, int height)
-{
-    AVFrame *picture;
-    int ret;
+static AVFrame *frame;
+static AVPicture src_picture, dst_picture;
+static int frame_count;
 
-    picture = av_frame_alloc();
-    if (!picture)
-        return NULL;
-
-    picture->format = pix_fmt;
-    picture->width  = width;
-    picture->height = height;
-
-    /* allocate the buffers for the frame data */
-    ret = av_frame_get_buffer(picture, 32);
-    if (ret < 0) {
-        fprintf(stderr, "Could not allocate frame data.\n");
-        exit(1);
-    }
-
-    return picture;
-}
-
-static void open_video(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
+static void open_video(AVFormatContext *oc, AVCodec *codec, AVStream *st)
 {
     int ret;
-    AVCodecContext *c = ost->st->codec;
-    AVDictionary *opt = NULL;
-
-    av_dict_copy(&opt, opt_arg, 0);
+    AVCodecContext *c = st->codec;
 
     /* open the codec */
-    ret = avcodec_open2(c, codec, &opt);
-    av_dict_free(&opt);
+    ret = avcodec_open2(c, codec, NULL);
     if (ret < 0) {
         fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret));
         exit(1);
     }
 
     /* allocate and init a re-usable frame */
-    ost->frame = alloc_picture(c->pix_fmt, c->width, c->height);
-    if (!ost->frame) {
+    frame = avcodec_alloc_frame();
+    if (!frame) {
         fprintf(stderr, "Could not allocate video frame\n");
         exit(1);
     }
 
+    /* Allocate the encoded raw picture. */
+    ret = avpicture_alloc(&dst_picture, c->pix_fmt, c->width, c->height);
+    if (ret < 0) {
+        fprintf(stderr, "Could not allocate picture: %s\n", av_err2str(ret));
+        exit(1);
+    }
+
     /* If the output format is not YUV420P, then a temporary YUV420P
      * picture is needed too. It is then converted to the required
      * output format. */
-    ost->tmp_frame = NULL;
     if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
-        ost->tmp_frame = alloc_picture(AV_PIX_FMT_YUV420P, c->width, c->height);
-        if (!ost->tmp_frame) {
-            fprintf(stderr, "Could not allocate temporary picture\n");
+        ret = avpicture_alloc(&src_picture, AV_PIX_FMT_YUV420P, c->width, c->height);
+        if (ret < 0) {
+            fprintf(stderr, "Could not allocate temporary picture: %s\n",
+                    av_err2str(ret));
             exit(1);
         }
     }
+
+    /* copy data and linesize picture pointers to frame */
+    *((AVPicture *)frame) = dst_picture;
 }
 
 /* Prepare a dummy image. */
-static void fill_yuv_image(AVFrame *pict, int frame_index,
+static void fill_yuv_image(AVPicture *pict, int frame_index,
                            int width, int height)
 {
-    int x, y, i, ret;
-
-    /* when we pass a frame to the encoder, it may keep a reference to it
-     * internally;
-     * make sure we do not overwrite it here
-     */
-    ret = av_frame_make_writable(pict);
-    if (ret < 0)
-        exit(1);
+    int x, y, i;
 
     i = frame_index;
 
@@ -446,77 +361,53 @@ static void fill_yuv_image(AVFrame *pict, int frame_index,
     }
 }
 
-static AVFrame *get_video_frame(OutputStream *ost)
+static void write_video_frame(AVFormatContext *oc, AVStream *st)
 {
-    AVCodecContext *c = ost->st->codec;
-
-    /* check if we want to generate more frames */
-    if (av_compare_ts(ost->next_pts, ost->st->codec->time_base,
-                      STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
-        return NULL;
+    int ret;
+    static struct SwsContext *sws_ctx;
+    AVCodecContext *c = st->codec;
 
+    if (frame_count >= STREAM_NB_FRAMES) {
+        /* No more frames to compress. The codec has a latency of a few
+         * frames if using B-frames, so we get the last frames by
+         * passing the same picture again. */
+    } else {
         if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
             /* as we only generate a YUV420P picture, we must convert it
              * to the codec pixel format if needed */
-        if (!ost->sws_ctx) {
-            ost->sws_ctx = sws_getContext(c->width, c->height,
-                                          AV_PIX_FMT_YUV420P,
-                                          c->width, c->height,
-                                          c->pix_fmt,
-                                          SCALE_FLAGS, NULL, NULL, NULL);
-            if (!ost->sws_ctx) {
+            if (!sws_ctx) {
+                sws_ctx = sws_getContext(c->width, c->height, AV_PIX_FMT_YUV420P,
+                                         c->width, c->height, c->pix_fmt,
+                                         sws_flags, NULL, NULL, NULL);
+                if (!sws_ctx) {
                     fprintf(stderr,
                             "Could not initialize the conversion context\n");
                     exit(1);
                 }
             }
-        fill_yuv_image(ost->tmp_frame, ost->next_pts, c->width, c->height);
-        sws_scale(ost->sws_ctx,
-                  (const uint8_t * const *)ost->tmp_frame->data, ost->tmp_frame->linesize,
-                  0, c->height, ost->frame->data, ost->frame->linesize);
+            fill_yuv_image(&src_picture, frame_count, c->width, c->height);
+            sws_scale(sws_ctx,
+                      (const uint8_t * const *)src_picture.data, src_picture.linesize,
+                      0, c->height, dst_picture.data, dst_picture.linesize);
         } else {
-        fill_yuv_image(ost->frame, ost->next_pts, c->width, c->height);
+            fill_yuv_image(&dst_picture, frame_count, c->width, c->height);
         }
-
-    ost->frame->pts = ost->next_pts++;
-
-    return ost->frame;
     }
 
-/*
- * encode one video frame and send it to the muxer
- * return 1 when encoding is finished, 0 otherwise
- */
-static int write_video_frame(AVFormatContext *oc, OutputStream *ost)
-{
-    int ret;
-    AVCodecContext *c;
-    AVFrame *frame;
-    int got_packet = 0;
-
-    c = ost->st->codec;
-
-    frame = get_video_frame(ost);
-
     if (oc->oformat->flags & AVFMT_RAWPICTURE) {
-        /* a hack to avoid data copy with some raw video muxers */
+        /* Raw video case - directly store the picture in the packet */
         AVPacket pkt;
         av_init_packet(&pkt);
 
-        if (!frame)
-            return 1;
-
         pkt.flags        |= AV_PKT_FLAG_KEY;
-        pkt.stream_index  = ost->st->index;
-        pkt.data          = (uint8_t *)frame;
+        pkt.stream_index  = st->index;
+        pkt.data          = dst_picture.data[0];
         pkt.size          = sizeof(AVPicture);
 
-        pkt.pts = pkt.dts = frame->pts;
-        av_packet_rescale_ts(&pkt, c->time_base, ost->st->time_base);
-
         ret = av_interleaved_write_frame(oc, &pkt);
     } else {
         AVPacket pkt = { 0 };
+        int got_packet;
         av_init_packet(&pkt);
 
         /* encode the image */
@@ -525,29 +416,30 @@ static int write_video_frame(AVFormatContext *oc, OutputStream *ost)
             fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret));
             exit(1);
         }
+        /* If size is zero, it means the image was buffered. */
 
-        if (got_packet) {
-            ret = write_frame(oc, &c->time_base, ost->st, &pkt);
+        if (!ret && got_packet && pkt.size) {
+            pkt.stream_index = st->index;
+
+            /* Write the compressed frame to the media file. */
+            ret = av_interleaved_write_frame(oc, &pkt);
         } else {
             ret = 0;
         }
     }
-
-    if (ret < 0) {
+    if (ret != 0) {
         fprintf(stderr, "Error while writing video frame: %s\n", av_err2str(ret));
         exit(1);
     }
-
-    return (frame || got_packet) ? 0 : 1;
+    frame_count++;
 }
 
-static void close_stream(AVFormatContext *oc, OutputStream *ost)
+static void close_video(AVFormatContext *oc, AVStream *st)
 {
-    avcodec_close(ost->st->codec);
-    av_frame_free(&ost->frame);
-    av_frame_free(&ost->tmp_frame);
-    sws_freeContext(ost->sws_ctx);
-    swr_free(&ost->swr_ctx);
+    avcodec_close(st->codec);
+    av_free(src_picture.data[0]);
+    av_free(dst_picture.data[0]);
+    av_free(frame);
 }
 
 /**************************************************************/
@@ -555,20 +447,18 @@ static void close_stream(AVFormatContext *oc, OutputStream *ost)
 
 int main(int argc, char **argv)
 {
-    OutputStream video_st = { 0 }, audio_st = { 0 };
     const char *filename;
     AVOutputFormat *fmt;
     AVFormatContext *oc;
+    AVStream *audio_st, *video_st;
     AVCodec *audio_codec, *video_codec;
+    double audio_time, video_time;
     int ret;
-    int have_video = 0, have_audio = 0;
-    int encode_video = 0, encode_audio = 0;
-    AVDictionary *opt = NULL;
 
     /* Initialize libavcodec, and register all codecs and formats. */
     av_register_all();
 
-    if (argc < 2) {
+    if (argc != 2) {
         printf("usage: %s output_file\n"
                "API example program to output a media file with libavformat.\n"
                "This program generates a synthetic audio and video stream, encodes and\n"
@@ -580,9 +470,6 @@ int main(int argc, char **argv)
     }
 
     filename = argv[1];
-    if (argc > 3 && !strcmp(argv[2], "-flags")) {
-        av_dict_set(&opt, argv[2]+1, argv[3], 0);
-    }
 
     /* allocate the output media context */
     avformat_alloc_output_context2(&oc, NULL, NULL, filename);
@@ -590,31 +477,29 @@ int main(int argc, char **argv)
         printf("Could not deduce output format from file extension: using MPEG.\n");
         avformat_alloc_output_context2(&oc, NULL, "mpeg", filename);
     }
-    if (!oc)
+    if (!oc) {
         return 1;
-
+    }
     fmt = oc->oformat;
 
     /* Add the audio and video streams using the default format codecs
      * and initialize the codecs. */
+    video_st = NULL;
+    audio_st = NULL;
+
     if (fmt->video_codec != AV_CODEC_ID_NONE) {
-        add_stream(&video_st, oc, &video_codec, fmt->video_codec);
-        have_video = 1;
-        encode_video = 1;
+        video_st = add_stream(oc, &video_codec, fmt->video_codec);
     }
     if (fmt->audio_codec != AV_CODEC_ID_NONE) {
-        add_stream(&audio_st, oc, &audio_codec, fmt->audio_codec);
-        have_audio = 1;
-        encode_audio = 1;
+        audio_st = add_stream(oc, &audio_codec, fmt->audio_codec);
     }
 
     /* Now that all the parameters are set, we can open the audio and
      * video codecs and allocate the necessary encode buffers. */
-    if (have_video)
-        open_video(oc, video_codec, &video_st, opt);
-
-    if (have_audio)
-        open_audio(oc, audio_codec, &audio_st, opt);
+    if (video_st)
+        open_video(oc, video_codec, video_st);
+    if (audio_st)
+        open_audio(oc, audio_codec, audio_st);
 
     av_dump_format(oc, 0, filename, 1);
 
@@ -629,21 +514,30 @@ int main(int argc, char **argv)
     }
 
     /* Write the stream header, if any. */
-    ret = avformat_write_header(oc, &opt);
+    ret = avformat_write_header(oc, NULL);
     if (ret < 0) {
         fprintf(stderr, "Error occurred when opening output file: %s\n",
                 av_err2str(ret));
         return 1;
     }
 
-    while (encode_video || encode_audio) {
-        /* select the stream to encode */
-        if (encode_video &&
-            (!encode_audio || av_compare_ts(video_st.next_pts, video_st.st->codec->time_base,
-                                            audio_st.next_pts, audio_st.st->codec->time_base) <= 0)) {
-            encode_video = !write_video_frame(oc, &video_st);
+    if (frame)
+        frame->pts = 0;
+    for (;;) {
+        /* Compute current audio and video time. */
+        audio_time = audio_st ? audio_st->pts.val * av_q2d(audio_st->time_base) : 0.0;
+        video_time = video_st ? video_st->pts.val * av_q2d(video_st->time_base) : 0.0;
+
+        if ((!audio_st || audio_time >= STREAM_DURATION) &&
+            (!video_st || video_time >= STREAM_DURATION))
+            break;
+
+        /* write interleaved audio and video frames */
+        if (!video_st || (video_st && audio_st && audio_time < video_time)) {
+            write_audio_frame(oc, audio_st);
         } else {
-            encode_audio = !write_audio_frame(oc, &audio_st);
+            write_video_frame(oc, video_st);
+            frame->pts += av_rescale_q(1, video_st->codec->time_base, video_st->time_base);
         }
     }
 
@@ -654,14 +548,14 @@ int main(int argc, char **argv)
     av_write_trailer(oc);
 
     /* Close each codec. */
-    if (have_video)
-        close_stream(oc, &video_st);
-    if (have_audio)
-        close_stream(oc, &audio_st);
+    if (video_st)
+        close_video(oc, video_st);
+    if (audio_st)
+        close_audio(oc, audio_st);
 
     if (!(fmt->flags & AVFMT_NOFILE))
         /* Close the output file. */
-        avio_closep(&oc->pb);
+        avio_close(oc->pb);
 
     /* free the stream */
     avformat_free_context(oc);

doc/examples/qsvdec.c

@@ -1,484 +0,0 @@
-/*
- * Copyright (c) 2015 Anton Khirnov
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
- * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- * THE SOFTWARE.
- */
-
-/**
- * @file
- * Intel QSV-accelerated H.264 decoding example.
- *
- * @example qsvdec.c
- * This example shows how to do QSV-accelerated H.264 decoding with output
- * frames in the VA-API video surfaces.
- */
-
-#include "config.h"
-
-#include <stdio.h>
-
-#include <mfx/mfxvideo.h>
-
-#include <va/va.h>
-#include <va/va_x11.h>
-#include <X11/Xlib.h>
-
-#include "libavformat/avformat.h"
-#include "libavformat/avio.h"
-
-#include "libavcodec/avcodec.h"
-#include "libavcodec/qsv.h"
-
-#include "libavutil/error.h"
-#include "libavutil/mem.h"
-
-typedef struct DecodeContext {
-    mfxSession mfx_session;
-    VADisplay va_dpy;
-
-    VASurfaceID *surfaces;
-    mfxMemId    *surface_ids;
-    int         *surface_used;
-    int       nb_surfaces;
-
-    mfxFrameInfo frame_info;
-} DecodeContext;
-
-static mfxStatus frame_alloc(mfxHDL pthis, mfxFrameAllocRequest *req,
-                             mfxFrameAllocResponse *resp)
-{
-    DecodeContext *decode = pthis;
-    int err, i;
-
-    if (decode->surfaces) {
-        fprintf(stderr, "Multiple allocation requests.\n");
-        return MFX_ERR_MEMORY_ALLOC;
-    }
-    if (!(req->Type & MFX_MEMTYPE_VIDEO_MEMORY_DECODER_TARGET)) {
-        fprintf(stderr, "Unsupported surface type: %d\n", req->Type);
-        return MFX_ERR_UNSUPPORTED;
-    }
-    if (req->Info.BitDepthLuma != 8 || req->Info.BitDepthChroma != 8 ||
-        req->Info.Shift || req->Info.FourCC != MFX_FOURCC_NV12 ||
-        req->Info.ChromaFormat != MFX_CHROMAFORMAT_YUV420) {
-        fprintf(stderr, "Unsupported surface properties.\n");
-        return MFX_ERR_UNSUPPORTED;
-    }
-
-    decode->surfaces     = av_malloc_array (req->NumFrameSuggested, sizeof(*decode->surfaces));
-    decode->surface_ids  = av_malloc_array (req->NumFrameSuggested, sizeof(*decode->surface_ids));
-    decode->surface_used = av_mallocz_array(req->NumFrameSuggested, sizeof(*decode->surface_used));
-    if (!decode->surfaces || !decode->surface_ids || !decode->surface_used)
-        goto fail;
-
-    err = vaCreateSurfaces(decode->va_dpy, VA_RT_FORMAT_YUV420,
-                           req->Info.Width, req->Info.Height,
-                           decode->surfaces, req->NumFrameSuggested,
-                           NULL, 0);
-    if (err != VA_STATUS_SUCCESS) {
-        fprintf(stderr, "Error allocating VA surfaces\n");
-        goto fail;
-    }
-    decode->nb_surfaces = req->NumFrameSuggested;
-
-    for (i = 0; i < decode->nb_surfaces; i++)
-        decode->surface_ids[i] = &decode->surfaces[i];
-
-    resp->mids           = decode->surface_ids;
-    resp->NumFrameActual = decode->nb_surfaces;
-
-    decode->frame_info = req->Info;
-
-    return MFX_ERR_NONE;
-fail:
-    av_freep(&decode->surfaces);
-    av_freep(&decode->surface_ids);
-    av_freep(&decode->surface_used);
-
-    return MFX_ERR_MEMORY_ALLOC;
-}
-
-static mfxStatus frame_free(mfxHDL pthis, mfxFrameAllocResponse *resp)
-{
-    DecodeContext *decode = pthis;
-
-    if (decode->surfaces)
-        vaDestroySurfaces(decode->va_dpy, decode->surfaces, decode->nb_surfaces);
-    av_freep(&decode->surfaces);
-    av_freep(&decode->surface_ids);
-    av_freep(&decode->surface_used);
-    decode->nb_surfaces = 0;
-
-    return MFX_ERR_NONE;
-}
-
-static mfxStatus frame_lock(mfxHDL pthis, mfxMemId mid, mfxFrameData *ptr)
-{
-    return MFX_ERR_UNSUPPORTED;
-}
-
-static mfxStatus frame_unlock(mfxHDL pthis, mfxMemId mid, mfxFrameData *ptr)
-{
-    return MFX_ERR_UNSUPPORTED;
-}
-
-static mfxStatus frame_get_hdl(mfxHDL pthis, mfxMemId mid, mfxHDL *hdl)
-{
-    *hdl = mid;
-    return MFX_ERR_NONE;
-}
-
-static void free_buffer(void *opaque, uint8_t *data)
-{
-    int *used = opaque;
-    *used = 0;
-    av_freep(&data);
-}
-
-static int get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
-{
-    DecodeContext *decode = avctx->opaque;
-
-    mfxFrameSurface1 *surf;
-    AVBufferRef *surf_buf;
-    int idx;
-
-    for (idx = 0; idx < decode->nb_surfaces; idx++) {
-        if (!decode->surface_used[idx])
-            break;
-    }
-    if (idx == decode->nb_surfaces) {
-        fprintf(stderr, "No free surfaces\n");
-        return AVERROR(ENOMEM);
-    }
-
-    surf = av_mallocz(sizeof(*surf));
-    if (!surf)
-        return AVERROR(ENOMEM);
-    surf_buf = av_buffer_create((uint8_t*)surf, sizeof(*surf), free_buffer,
-                                &decode->surface_used[idx], AV_BUFFER_FLAG_READONLY);
-    if (!surf_buf) {
-        av_freep(&surf);
-        return AVERROR(ENOMEM);
-    }
-
-    surf->Info       = decode->frame_info;
-    surf->Data.MemId = &decode->surfaces[idx];
-
-    frame->buf[0]  = surf_buf;
-    frame->data[3] = (uint8_t*)surf;
-
-    decode->surface_used[idx] = 1;
-
-    return 0;
-}
-
-static int get_format(AVCodecContext *avctx, const enum AVPixelFormat *pix_fmts)
-{
-    while (*pix_fmts != AV_PIX_FMT_NONE) {
-        if (*pix_fmts == AV_PIX_FMT_QSV) {
-            if (!avctx->hwaccel_context) {
-                DecodeContext *decode = avctx->opaque;
-                AVQSVContext *qsv = av_qsv_alloc_context();
-                if (!qsv)
-                    return AV_PIX_FMT_NONE;
-
-                qsv->session   = decode->mfx_session;
-                qsv->iopattern = MFX_IOPATTERN_OUT_VIDEO_MEMORY;
-
-                avctx->hwaccel_context = qsv;
-            }
-
-            return AV_PIX_FMT_QSV;
-        }
-
-        pix_fmts++;
-    }
-
-    fprintf(stderr, "The QSV pixel format not offered in get_format()\n");
-
-    return AV_PIX_FMT_NONE;
-}
-
-static int decode_packet(DecodeContext *decode, AVCodecContext *decoder_ctx,
-                         AVFrame *frame, AVPacket *pkt,
-                         AVIOContext *output_ctx)
-{
-    int ret = 0;
-    int got_frame = 1;
-
-    while (pkt->size > 0 || (!pkt->data && got_frame)) {
-        ret = avcodec_decode_video2(decoder_ctx, frame, &got_frame, pkt);
-        if (ret < 0) {
-            fprintf(stderr, "Error during decoding\n");
-            return ret;
-        }
-
-        pkt->data += ret;
-        pkt->size -= ret;
-
-        /* A real program would do something useful with the decoded frame here.
-         * We just retrieve the raw data and write it to a file, which is rather
-         * useless but pedagogic. */
-        if (got_frame) {
-            mfxFrameSurface1 *surf = (mfxFrameSurface1*)frame->data[3];
-            VASurfaceID    surface = *(VASurfaceID*)surf->Data.MemId;
-
-            VAImageFormat img_fmt = {
-                .fourcc         = VA_FOURCC_NV12,
-                .byte_order     = VA_LSB_FIRST,
-                .bits_per_pixel = 8,
-                .depth          = 8,
-            };
-
-            VAImage img;
-
-            VAStatus err;
-            uint8_t *data;
-            int i, j;
-
-            img.buf      = VA_INVALID_ID;
-            img.image_id = VA_INVALID_ID;
-
-            err = vaCreateImage(decode->va_dpy, &img_fmt,
-                                frame->width, frame->height, &img);
-            if (err != VA_STATUS_SUCCESS) {
-                fprintf(stderr, "Error creating an image: %s\n",
-                        vaErrorStr(err));
-                ret = AVERROR_UNKNOWN;
-                goto fail;
-            }
-
-            err = vaGetImage(decode->va_dpy, surface, 0, 0,
-                             frame->width, frame->height,
-                             img.image_id);
-            if (err != VA_STATUS_SUCCESS) {
-                fprintf(stderr, "Error getting an image: %s\n",
-                        vaErrorStr(err));
-                ret = AVERROR_UNKNOWN;
-                goto fail;
-            }
-
-            err = vaMapBuffer(decode->va_dpy, img.buf, (void**)&data);
-            if (err != VA_STATUS_SUCCESS) {
-                fprintf(stderr, "Error mapping the image buffer: %s\n",
-                        vaErrorStr(err));
-                ret = AVERROR_UNKNOWN;
-                goto fail;
-            }
-
-            for (i = 0; i < img.num_planes; i++)
-                for (j = 0; j < (img.height >> (i > 0)); j++)
-                    avio_write(output_ctx, data + img.offsets[i] + j * img.pitches[i], img.width);
-
-fail:
-            if (img.buf != VA_INVALID_ID)
-                vaUnmapBuffer(decode->va_dpy, img.buf);
-            if (img.image_id != VA_INVALID_ID)
-                vaDestroyImage(decode->va_dpy, img.image_id);
-            av_frame_unref(frame);
-
-            if (ret < 0)
-                return ret;
-        }
-    }
-
-    return 0;
-}
-
-int main(int argc, char **argv)
-{
-    AVFormatContext *input_ctx = NULL;
-    AVStream *video_st = NULL;
-    AVCodecContext *decoder_ctx = NULL;
-    const AVCodec *decoder;
-
-    AVPacket pkt = { 0 };
-    AVFrame *frame = NULL;
-
-    DecodeContext decode = { NULL };
-
-    Display *dpy = NULL;
-    int va_ver_major, va_ver_minor;
-
-    mfxIMPL mfx_impl = MFX_IMPL_AUTO_ANY;
-    mfxVersion mfx_ver = { { 1, 1 } };
-
-    mfxFrameAllocator frame_allocator = {
-        .pthis = &decode,
-        .Alloc = frame_alloc,
-        .Lock  = frame_lock,
-        .Unlock = frame_unlock,
-        .GetHDL = frame_get_hdl,
-        .Free   = frame_free,
-    };
-
-    AVIOContext *output_ctx = NULL;
-
-    int ret, i, err;
-
-    av_register_all();
-
-    if (argc < 3) {
-        fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
-        return 1;
-    }
-
-    /* open the input file */
-    ret = avformat_open_input(&input_ctx, argv[1], NULL, NULL);
-    if (ret < 0) {
-        fprintf(stderr, "Cannot open input file '%s': ", argv[1]);
-        goto finish;
-    }
-
-    /* find the first H.264 video stream */
-    for (i = 0; i < input_ctx->nb_streams; i++) {
-        AVStream *st = input_ctx->streams[i];
-
-        if (st->codec->codec_id == AV_CODEC_ID_H264 && !video_st)
-            video_st = st;
-        else
-            st->discard = AVDISCARD_ALL;
-    }
-    if (!video_st) {
-        fprintf(stderr, "No H.264 video stream in the input file\n");
-        goto finish;
-    }
-
-    /* initialize VA-API */
-    dpy = XOpenDisplay(NULL);
-    if (!dpy) {
-        fprintf(stderr, "Cannot open the X display\n");
-        goto finish;
-    }
-    decode.va_dpy = vaGetDisplay(dpy);
-    if (!decode.va_dpy) {
-        fprintf(stderr, "Cannot open the VA display\n");
-        goto finish;
-    }
-
-    err = vaInitialize(decode.va_dpy, &va_ver_major, &va_ver_minor);
-    if (err != VA_STATUS_SUCCESS) {
-        fprintf(stderr, "Cannot initialize VA: %s\n", vaErrorStr(err));
-        goto finish;
-    }
-    fprintf(stderr, "Initialized VA v%d.%d\n", va_ver_major, va_ver_minor);
-
-    /* initialize an MFX session */
-    err = MFXInit(mfx_impl, &mfx_ver, &decode.mfx_session);
-    if (err != MFX_ERR_NONE) {
-        fprintf(stderr, "Error initializing an MFX session\n");
-        goto finish;
-    }
-
-    MFXVideoCORE_SetHandle(decode.mfx_session, MFX_HANDLE_VA_DISPLAY, decode.va_dpy);
-    MFXVideoCORE_SetFrameAllocator(decode.mfx_session, &frame_allocator);
-
-    /* initialize the decoder */
-    decoder = avcodec_find_decoder_by_name("h264_qsv");
-    if (!decoder) {
-        fprintf(stderr, "The QSV decoder is not present in libavcodec\n");
-        goto finish;
-    }
-
-    decoder_ctx = avcodec_alloc_context3(decoder);
-    if (!decoder_ctx) {
-        ret = AVERROR(ENOMEM);
-        goto finish;
-    }
-    decoder_ctx->codec_id = AV_CODEC_ID_H264;
-    if (video_st->codec->extradata_size) {
-        decoder_ctx->extradata = av_mallocz(video_st->codec->extradata_size +
-                                            FF_INPUT_BUFFER_PADDING_SIZE);
-        if (!decoder_ctx->extradata) {
-            ret = AVERROR(ENOMEM);
-            goto finish;
-        }
-        memcpy(decoder_ctx->extradata, video_st->codec->extradata,
-               video_st->codec->extradata_size);
-        decoder_ctx->extradata_size = video_st->codec->extradata_size;
-    }
-    decoder_ctx->refcounted_frames = 1;
-
-    decoder_ctx->opaque      = &decode;
-    decoder_ctx->get_buffer2 = get_buffer;
-    decoder_ctx->get_format  = get_format;
-
-    ret = avcodec_open2(decoder_ctx, NULL, NULL);
-    if (ret < 0) {
-        fprintf(stderr, "Error opening the decoder: ");
-        goto finish;
-    }
-
-    /* open the output stream */
-    ret = avio_open(&output_ctx, argv[2], AVIO_FLAG_WRITE);
-    if (ret < 0) {
-        fprintf(stderr, "Error opening the output context: ");
-        goto finish;
-    }
-
-    frame = av_frame_alloc();
-    if (!frame) {
-        ret = AVERROR(ENOMEM);
-        goto finish;
-    }
-
-    /* actual decoding */
-    while (ret >= 0) {
-        ret = av_read_frame(input_ctx, &pkt);
-        if (ret < 0)
-            break;
-
-        if (pkt.stream_index == video_st->index)
-            ret = decode_packet(&decode, decoder_ctx, frame, &pkt, output_ctx);
-
-        av_packet_unref(&pkt);
-    }
-
-    /* flush the decoder */
-    pkt.data = NULL;
-    pkt.size = 0;
-    ret = decode_packet(&decode, decoder_ctx, frame, &pkt, output_ctx);
-
-finish:
-    if (ret < 0) {
-        char buf[1024];
-        av_strerror(ret, buf, sizeof(buf));
-        fprintf(stderr, "%s\n", buf);
-    }
-
-    avformat_close_input(&input_ctx);
-
-    av_frame_free(&frame);
-
-    if (decode.mfx_session)
-        MFXClose(decode.mfx_session);
-    if (decode.va_dpy)
-        vaTerminate(decode.va_dpy);
-    if (dpy)
-        XCloseDisplay(dpy);
-
-    if (decoder_ctx)
-        av_freep(&decoder_ctx->hwaccel_context);
-    avcodec_free_context(&decoder_ctx);
-
-    avio_close(output_ctx);
-
-    return ret;
-}

doc/examples/remuxing.c

@@ -1,165 +0,0 @@
-/*
- * Copyright (c) 2013 Stefano Sabatini
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
- * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- * THE SOFTWARE.
- */
-
-/**
- * @file
- * libavformat/libavcodec demuxing and muxing API example.
- *
- * Remux streams from one container format to another.
- * @example remuxing.c
- */
-
-#include <libavutil/timestamp.h>
-#include <libavformat/avformat.h>
-
-static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt, const char *tag)
-{
-    AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
-
-    printf("%s: pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
-           tag,
-           av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base),
-           av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base),
-           av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base),
-           pkt->stream_index);
-}
-
-int main(int argc, char **argv)
-{
-    AVOutputFormat *ofmt = NULL;
-    AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL;
-    AVPacket pkt;
-    const char *in_filename, *out_filename;
-    int ret, i;
-
-    if (argc < 3) {
-        printf("usage: %s input output\n"
-               "API example program to remux a media file with libavformat and libavcodec.\n"
-               "The output format is guessed according to the file extension.\n"
-               "\n", argv[0]);
-        return 1;
-    }
-
-    in_filename  = argv[1];
-    out_filename = argv[2];
-
-    av_register_all();
-
-    if ((ret = avformat_open_input(&ifmt_ctx, in_filename, 0, 0)) < 0) {
-        fprintf(stderr, "Could not open input file '%s'", in_filename);
-        goto end;
-    }
-
-    if ((ret = avformat_find_stream_info(ifmt_ctx, 0)) < 0) {
-        fprintf(stderr, "Failed to retrieve input stream information");
-        goto end;
-    }
-
-    av_dump_format(ifmt_ctx, 0, in_filename, 0);
-
-    avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, out_filename);
-    if (!ofmt_ctx) {
-        fprintf(stderr, "Could not create output context\n");
-        ret = AVERROR_UNKNOWN;
-        goto end;
-    }
-
-    ofmt = ofmt_ctx->oformat;
-
-    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
-        AVStream *in_stream = ifmt_ctx->streams[i];
-        AVStream *out_stream = avformat_new_stream(ofmt_ctx, in_stream->codec->codec);
-        if (!out_stream) {
-            fprintf(stderr, "Failed allocating output stream\n");
-            ret = AVERROR_UNKNOWN;
-            goto end;
-        }
-
-        ret = avcodec_copy_context(out_stream->codec, in_stream->codec);
-        if (ret < 0) {
-            fprintf(stderr, "Failed to copy context from input to output stream codec context\n");
-            goto end;
-        }
-        out_stream->codec->codec_tag = 0;
-        if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
-            out_stream->codec->flags |= CODEC_FLAG_GLOBAL_HEADER;
-    }
-    av_dump_format(ofmt_ctx, 0, out_filename, 1);
-
-    if (!(ofmt->flags & AVFMT_NOFILE)) {
-        ret = avio_open(&ofmt_ctx->pb, out_filename, AVIO_FLAG_WRITE);
-        if (ret < 0) {
-            fprintf(stderr, "Could not open output file '%s'", out_filename);
-            goto end;
-        }
-    }
-
-    ret = avformat_write_header(ofmt_ctx, NULL);
-    if (ret < 0) {
-        fprintf(stderr, "Error occurred when opening output file\n");
-        goto end;
-    }
-
-    while (1) {
-        AVStream *in_stream, *out_stream;
-
-        ret = av_read_frame(ifmt_ctx, &pkt);
-        if (ret < 0)
-            break;
-
-        in_stream  = ifmt_ctx->streams[pkt.stream_index];
-        out_stream = ofmt_ctx->streams[pkt.stream_index];
-
-        log_packet(ifmt_ctx, &pkt, "in");
-
-        /* copy packet */
-        pkt.pts = av_rescale_q_rnd(pkt.pts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
-        pkt.dts = av_rescale_q_rnd(pkt.dts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
-        pkt.duration = av_rescale_q(pkt.duration, in_stream->time_base, out_stream->time_base);
-        pkt.pos = -1;
-        log_packet(ofmt_ctx, &pkt, "out");
-
-        ret = av_interleaved_write_frame(ofmt_ctx, &pkt);
-        if (ret < 0) {
-            fprintf(stderr, "Error muxing packet\n");
-            break;
-        }
-        av_free_packet(&pkt);
-    }
-
-    av_write_trailer(ofmt_ctx);
-end:
-
-    avformat_close_input(&ifmt_ctx);
-
-    /* close output */
-    if (ofmt_ctx && !(ofmt->flags & AVFMT_NOFILE))
-        avio_closep(&ofmt_ctx->pb);
-    avformat_free_context(ofmt_ctx);
-
-    if (ret < 0 && ret != AVERROR_EOF) {
-        fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
-        return 1;
-    }
-
-    return 0;
-}

doc/examples/resampling_audio.c

@@ -21,7 +21,7 @@
  */
 
 /**
- * @example resampling_audio.c
+ * @example doc/examples/resampling_audio.c
  * libswresample API use example.
  */
 
@@ -62,7 +62,7 @@ static int get_format_from_sample_fmt(const char **fmt,
 /**
  * Fill dst buffer with nb_samples, generated starting from t.
  */
-static void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
+void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
 {
     int i, j;
     double tincr = 1.0 / sample_rate, *dstp = dst;
@@ -168,7 +168,7 @@ int main(int argc, char **argv)
         dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +
                                         src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
         if (dst_nb_samples > max_dst_nb_samples) {
-            av_freep(&dst_data[0]);
+            av_free(dst_data[0]);
             ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
                                    dst_nb_samples, dst_sample_fmt, 1);
             if (ret < 0)
@@ -184,10 +184,6 @@ int main(int argc, char **argv)
         }
         dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
                                                  ret, dst_sample_fmt, 1);
-        if (dst_bufsize < 0) {
-            fprintf(stderr, "Could not get sample buffer size\n");
-            goto end;
-        }
         printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);
         fwrite(dst_data[0], 1, dst_bufsize, dst_file);
     } while (t < 10);
@@ -199,6 +195,7 @@ int main(int argc, char **argv)
             fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);
 
 end:
+    if (dst_file)
         fclose(dst_file);
 
     if (src_data)

doc/examples/scaling_video.c

@@ -23,7 +23,7 @@
 /**
  * @file
  * libswscale API use example.
- * @example scaling_video.c
+ * @example doc/examples/scaling_video.c
  */
 
 #include <libavutil/imgutils.h>
@@ -132,6 +132,7 @@ int main(int argc, char **argv)
            av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h, dst_filename);
 
 end:
+    if (dst_file)
         fclose(dst_file);
     av_freep(&src_data[0]);
     av_freep(&dst_data[0]);

doc/examples/transcode_aac.c

@@ -1,770 +0,0 @@
-/*
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * @file
- * simple audio converter
- *
- * @example transcode_aac.c
- * Convert an input audio file to AAC in an MP4 container using FFmpeg.
- * @author Andreas Unterweger (dustsigns@gmail.com)
- */
-
-#include <stdio.h>
-
-#include "libavformat/avformat.h"
-#include "libavformat/avio.h"
-
-#include "libavcodec/avcodec.h"
-
-#include "libavutil/audio_fifo.h"
-#include "libavutil/avassert.h"
-#include "libavutil/avstring.h"
-#include "libavutil/frame.h"
-#include "libavutil/opt.h"
-
-#include "libswresample/swresample.h"
-
-/** The output bit rate in kbit/s */
-#define OUTPUT_BIT_RATE 96000
-/** The number of output channels */
-#define OUTPUT_CHANNELS 2
-
-/**
- * Convert an error code into a text message.
- * @param error Error code to be converted
- * @return Corresponding error text (not thread-safe)
- */
-static const char *get_error_text(const int error)
-{
-    static char error_buffer[255];
-    av_strerror(error, error_buffer, sizeof(error_buffer));
-    return error_buffer;
-}
-
-/** Open an input file and the required decoder. */
-static int open_input_file(const char *filename,
-                           AVFormatContext **input_format_context,
-                           AVCodecContext **input_codec_context)
-{
-    AVCodec *input_codec;
-    int error;
-
-    /** Open the input file to read from it. */
-    if ((error = avformat_open_input(input_format_context, filename, NULL,
-                                     NULL)) < 0) {
-        fprintf(stderr, "Could not open input file '%s' (error '%s')\n",
-                filename, get_error_text(error));
-        *input_format_context = NULL;
-        return error;
-    }
-
-    /** Get information on the input file (number of streams etc.). */
-    if ((error = avformat_find_stream_info(*input_format_context, NULL)) < 0) {
-        fprintf(stderr, "Could not open find stream info (error '%s')\n",
-                get_error_text(error));
-        avformat_close_input(input_format_context);
-        return error;
-    }
-
-    /** Make sure that there is only one stream in the input file. */
-    if ((*input_format_context)->nb_streams != 1) {
-        fprintf(stderr, "Expected one audio input stream, but found %d\n",
-                (*input_format_context)->nb_streams);
-        avformat_close_input(input_format_context);
-        return AVERROR_EXIT;
-    }
-
-    /** Find a decoder for the audio stream. */
-    if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codec->codec_id))) {
-        fprintf(stderr, "Could not find input codec\n");
-        avformat_close_input(input_format_context);
-        return AVERROR_EXIT;
-    }
-
-    /** Open the decoder for the audio stream to use it later. */
-    if ((error = avcodec_open2((*input_format_context)->streams[0]->codec,
-                               input_codec, NULL)) < 0) {
-        fprintf(stderr, "Could not open input codec (error '%s')\n",
-                get_error_text(error));
-        avformat_close_input(input_format_context);
-        return error;
-    }
-
-    /** Save the decoder context for easier access later. */
-    *input_codec_context = (*input_format_context)->streams[0]->codec;
-
-    return 0;
-}
-
-/**
- * Open an output file and the required encoder.
- * Also set some basic encoder parameters.
- * Some of these parameters are based on the input file's parameters.
- */
-static int open_output_file(const char *filename,
-                            AVCodecContext *input_codec_context,
-                            AVFormatContext **output_format_context,
-                            AVCodecContext **output_codec_context)
-{
-    AVIOContext *output_io_context = NULL;
-    AVStream *stream               = NULL;
-    AVCodec *output_codec          = NULL;
-    int error;
-
-    /** Open the output file to write to it. */
-    if ((error = avio_open(&output_io_context, filename,
-                           AVIO_FLAG_WRITE)) < 0) {
-        fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
-                filename, get_error_text(error));
-        return error;
-    }
-
-    /** Create a new format context for the output container format. */
-    if (!(*output_format_context = avformat_alloc_context())) {
-        fprintf(stderr, "Could not allocate output format context\n");
-        return AVERROR(ENOMEM);
-    }
-
-    /** Associate the output file (pointer) with the container format context. */
-    (*output_format_context)->pb = output_io_context;
-
-    /** Guess the desired container format based on the file extension. */
-    if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
-                                                              NULL))) {
-        fprintf(stderr, "Could not find output file format\n");
-        goto cleanup;
-    }
-
-    av_strlcpy((*output_format_context)->filename, filename,
-               sizeof((*output_format_context)->filename));
-
-    /** Find the encoder to be used by its name. */
-    if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
-        fprintf(stderr, "Could not find an AAC encoder.\n");
-        goto cleanup;
-    }
-
-    /** Create a new audio stream in the output file container. */
-    if (!(stream = avformat_new_stream(*output_format_context, output_codec))) {
-        fprintf(stderr, "Could not create new stream\n");
-        error = AVERROR(ENOMEM);
-        goto cleanup;
-    }
-
-    /** Save the encoder context for easier access later. */
-    *output_codec_context = stream->codec;
-
-    /**
-     * Set the basic encoder parameters.
-     * The input file's sample rate is used to avoid a sample rate conversion.
-     */
-    (*output_codec_context)->channels       = OUTPUT_CHANNELS;
-    (*output_codec_context)->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
-    (*output_codec_context)->sample_rate    = input_codec_context->sample_rate;
-    (*output_codec_context)->sample_fmt     = output_codec->sample_fmts[0];
-    (*output_codec_context)->bit_rate       = OUTPUT_BIT_RATE;
-
-    /** Allow the use of the experimental AAC encoder */
-    (*output_codec_context)->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
-
-    /** Set the sample rate for the container. */
-    stream->time_base.den = input_codec_context->sample_rate;
-    stream->time_base.num = 1;
-
-    /**
-     * Some container formats (like MP4) require global headers to be present
-     * Mark the encoder so that it behaves accordingly.
-     */
-    if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
-        (*output_codec_context)->flags |= CODEC_FLAG_GLOBAL_HEADER;
-
-    /** Open the encoder for the audio stream to use it later. */
-    if ((error = avcodec_open2(*output_codec_context, output_codec, NULL)) < 0) {
-        fprintf(stderr, "Could not open output codec (error '%s')\n",
-                get_error_text(error));
-        goto cleanup;
-    }
-
-    return 0;
-
-cleanup:
-    avio_closep(&(*output_format_context)->pb);
-    avformat_free_context(*output_format_context);
-    *output_format_context = NULL;
-    return error < 0 ? error : AVERROR_EXIT;
-}
-
-/** Initialize one data packet for reading or writing. */
-static void init_packet(AVPacket *packet)
-{
-    av_init_packet(packet);
-    /** Set the packet data and size so that it is recognized as being empty. */
-    packet->data = NULL;
-    packet->size = 0;
-}
-
-/** Initialize one audio frame for reading from the input file */
-static int init_input_frame(AVFrame **frame)
-{
-    if (!(*frame = av_frame_alloc())) {
-        fprintf(stderr, "Could not allocate input frame\n");
-        return AVERROR(ENOMEM);
-    }
-    return 0;
-}
-
-/**
- * Initialize the audio resampler based on the input and output codec settings.
- * If the input and output sample formats differ, a conversion is required
- * libswresample takes care of this, but requires initialization.
- */
-static int init_resampler(AVCodecContext *input_codec_context,
-                          AVCodecContext *output_codec_context,
-                          SwrContext **resample_context)
-{
-        int error;
-
-        /**
-         * Create a resampler context for the conversion.
-         * Set the conversion parameters.
-         * Default channel layouts based on the number of channels
-         * are assumed for simplicity (they are sometimes not detected
-         * properly by the demuxer and/or decoder).
-         */
-        *resample_context = swr_alloc_set_opts(NULL,
-                                              av_get_default_channel_layout(output_codec_context->channels),
-                                              output_codec_context->sample_fmt,
-                                              output_codec_context->sample_rate,
-                                              av_get_default_channel_layout(input_codec_context->channels),
-                                              input_codec_context->sample_fmt,
-                                              input_codec_context->sample_rate,
-                                              0, NULL);
-        if (!*resample_context) {
-            fprintf(stderr, "Could not allocate resample context\n");
-            return AVERROR(ENOMEM);
-        }
-        /**
-        * Perform a sanity check so that the number of converted samples is
-        * not greater than the number of samples to be converted.
-        * If the sample rates differ, this case has to be handled differently
-        */
-        av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
-
-        /** Open the resampler with the specified parameters. */
-        if ((error = swr_init(*resample_context)) < 0) {
-            fprintf(stderr, "Could not open resample context\n");
-            swr_free(resample_context);
-            return error;
-        }
-    return 0;
-}
-
-/** Initialize a FIFO buffer for the audio samples to be encoded. */
-static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
-{
-    /** Create the FIFO buffer based on the specified output sample format. */
-    if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
-                                      output_codec_context->channels, 1))) {
-        fprintf(stderr, "Could not allocate FIFO\n");
-        return AVERROR(ENOMEM);
-    }
-    return 0;
-}
-
-/** Write the header of the output file container. */
-static int write_output_file_header(AVFormatContext *output_format_context)
-{
-    int error;
-    if ((error = avformat_write_header(output_format_context, NULL)) < 0) {
-        fprintf(stderr, "Could not write output file header (error '%s')\n",
-                get_error_text(error));
-        return error;
-    }
-    return 0;
-}
-
-/** Decode one audio frame from the input file. */
-static int decode_audio_frame(AVFrame *frame,
-                              AVFormatContext *input_format_context,
-                              AVCodecContext *input_codec_context,
-                              int *data_present, int *finished)
-{
-    /** Packet used for temporary storage. */
-    AVPacket input_packet;
-    int error;
-    init_packet(&input_packet);
-
-    /** Read one audio frame from the input file into a temporary packet. */
-    if ((error = av_read_frame(input_format_context, &input_packet)) < 0) {
-        /** If we are at the end of the file, flush the decoder below. */
-        if (error == AVERROR_EOF)
-            *finished = 1;
-        else {
-            fprintf(stderr, "Could not read frame (error '%s')\n",
-                    get_error_text(error));
-            return error;
-        }
-    }
-
-    /**
-     * Decode the audio frame stored in the temporary packet.
-     * The input audio stream decoder is used to do this.
-     * If we are at the end of the file, pass an empty packet to the decoder
-     * to flush it.
-     */
-    if ((error = avcodec_decode_audio4(input_codec_context, frame,
-                                       data_present, &input_packet)) < 0) {
-        fprintf(stderr, "Could not decode frame (error '%s')\n",
-                get_error_text(error));
-        av_free_packet(&input_packet);
-        return error;
-    }
-
-    /**
-     * If the decoder has not been flushed completely, we are not finished,
-     * so that this function has to be called again.
-     */
-    if (*finished && *data_present)
-        *finished = 0;
-    av_free_packet(&input_packet);
-    return 0;
-}
-
-/**
- * Initialize a temporary storage for the specified number of audio samples.
- * The conversion requires temporary storage due to the different format.
- * The number of audio samples to be allocated is specified in frame_size.
- */
-static int init_converted_samples(uint8_t ***converted_input_samples,
-                                  AVCodecContext *output_codec_context,
-                                  int frame_size)
-{
-    int error;
-
-    /**
-     * Allocate as many pointers as there are audio channels.
-     * Each pointer will later point to the audio samples of the corresponding
-     * channels (although it may be NULL for interleaved formats).
-     */
-    if (!(*converted_input_samples = calloc(output_codec_context->channels,
-                                            sizeof(**converted_input_samples)))) {
-        fprintf(stderr, "Could not allocate converted input sample pointers\n");
-        return AVERROR(ENOMEM);
-    }
-
-    /**
-     * Allocate memory for the samples of all channels in one consecutive
-     * block for convenience.
-     */
-    if ((error = av_samples_alloc(*converted_input_samples, NULL,
-                                  output_codec_context->channels,
-                                  frame_size,
-                                  output_codec_context->sample_fmt, 0)) < 0) {
-        fprintf(stderr,
-                "Could not allocate converted input samples (error '%s')\n",
-                get_error_text(error));
-        av_freep(&(*converted_input_samples)[0]);
-        free(*converted_input_samples);
-        return error;
-    }
-    return 0;
-}
-
-/**
- * Convert the input audio samples into the output sample format.
- * The conversion happens on a per-frame basis, the size of which is specified
- * by frame_size.
- */
-static int convert_samples(const uint8_t **input_data,
-                           uint8_t **converted_data, const int frame_size,
-                           SwrContext *resample_context)
-{
-    int error;
-
-    /** Convert the samples using the resampler. */
-    if ((error = swr_convert(resample_context,
-                             converted_data, frame_size,
-                             input_data    , frame_size)) < 0) {
-        fprintf(stderr, "Could not convert input samples (error '%s')\n",
-                get_error_text(error));
-        return error;
-    }
-
-    return 0;
-}
-
-/** Add converted input audio samples to the FIFO buffer for later processing. */
-static int add_samples_to_fifo(AVAudioFifo *fifo,
-                               uint8_t **converted_input_samples,
-                               const int frame_size)
-{
-    int error;
-
-    /**
-     * Make the FIFO as large as it needs to be to hold both,
-     * the old and the new samples.
-     */
-    if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
-        fprintf(stderr, "Could not reallocate FIFO\n");
-        return error;
-    }
-
-    /** Store the new samples in the FIFO buffer. */
-    if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
-                            frame_size) < frame_size) {
-        fprintf(stderr, "Could not write data to FIFO\n");
-        return AVERROR_EXIT;
-    }
-    return 0;
-}
-
-/**
- * Read one audio frame from the input file, decodes, converts and stores
- * it in the FIFO buffer.
- */
-static int read_decode_convert_and_store(AVAudioFifo *fifo,
-                                         AVFormatContext *input_format_context,
-                                         AVCodecContext *input_codec_context,
-                                         AVCodecContext *output_codec_context,
-                                         SwrContext *resampler_context,
-                                         int *finished)
-{
-    /** Temporary storage of the input samples of the frame read from the file. */
-    AVFrame *input_frame = NULL;
-    /** Temporary storage for the converted input samples. */
-    uint8_t **converted_input_samples = NULL;
-    int data_present;
-    int ret = AVERROR_EXIT;
-
-    /** Initialize temporary storage for one input frame. */
-    if (init_input_frame(&input_frame))
-        goto cleanup;
-    /** Decode one frame worth of audio samples. */
-    if (decode_audio_frame(input_frame, input_format_context,
-                           input_codec_context, &data_present, finished))
-        goto cleanup;
-    /**
-     * If we are at the end of the file and there are no more samples
-     * in the decoder which are delayed, we are actually finished.
-     * This must not be treated as an error.
-     */
-    if (*finished && !data_present) {
-        ret = 0;
-        goto cleanup;
-    }
-    /** If there is decoded data, convert and store it */
-    if (data_present) {
-        /** Initialize the temporary storage for the converted input samples. */
-        if (init_converted_samples(&converted_input_samples, output_codec_context,
-                                   input_frame->nb_samples))
-            goto cleanup;
-
-        /**
-         * Convert the input samples to the desired output sample format.
-         * This requires a temporary storage provided by converted_input_samples.
-         */
-        if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples,
-                            input_frame->nb_samples, resampler_context))
-            goto cleanup;
-
-        /** Add the converted input samples to the FIFO buffer for later processing. */
-        if (add_samples_to_fifo(fifo, converted_input_samples,
-                                input_frame->nb_samples))
-            goto cleanup;
-        ret = 0;
-    }
-    ret = 0;
-
-cleanup:
-    if (converted_input_samples) {
-        av_freep(&converted_input_samples[0]);
-        free(converted_input_samples);
-    }
-    av_frame_free(&input_frame);
-
-    return ret;
-}
-
-/**
- * Initialize one input frame for writing to the output file.
- * The frame will be exactly frame_size samples large.
- */
-static int init_output_frame(AVFrame **frame,
-                             AVCodecContext *output_codec_context,
-                             int frame_size)
-{
-    int error;
-
-    /** Create a new frame to store the audio samples. */
-    if (!(*frame = av_frame_alloc())) {
-        fprintf(stderr, "Could not allocate output frame\n");
-        return AVERROR_EXIT;
-    }
-
-    /**
-     * Set the frame's parameters, especially its size and format.
-     * av_frame_get_buffer needs this to allocate memory for the
-     * audio samples of the frame.
-     * Default channel layouts based on the number of channels
-     * are assumed for simplicity.
-     */
-    (*frame)->nb_samples     = frame_size;
-    (*frame)->channel_layout = output_codec_context->channel_layout;
-    (*frame)->format         = output_codec_context->sample_fmt;
-    (*frame)->sample_rate    = output_codec_context->sample_rate;
-
-    /**
-     * Allocate the samples of the created frame. This call will make
-     * sure that the audio frame can hold as many samples as specified.
-     */
-    if ((error = av_frame_get_buffer(*frame, 0)) < 0) {
-        fprintf(stderr, "Could allocate output frame samples (error '%s')\n",
-                get_error_text(error));
-        av_frame_free(frame);
-        return error;
-    }
-
-    return 0;
-}
-
-/** Global timestamp for the audio frames */
-static int64_t pts = 0;
-
-/** Encode one frame worth of audio to the output file. */
-static int encode_audio_frame(AVFrame *frame,
-                              AVFormatContext *output_format_context,
-                              AVCodecContext *output_codec_context,
-                              int *data_present)
-{
-    /** Packet used for temporary storage. */
-    AVPacket output_packet;
-    int error;
-    init_packet(&output_packet);
-
-    /** Set a timestamp based on the sample rate for the container. */
-    if (frame) {
-        frame->pts = pts;
-        pts += frame->nb_samples;
-    }
-
-    /**
-     * Encode the audio frame and store it in the temporary packet.
-     * The output audio stream encoder is used to do this.
-     */
-    if ((error = avcodec_encode_audio2(output_codec_context, &output_packet,
-                                       frame, data_present)) < 0) {
-        fprintf(stderr, "Could not encode frame (error '%s')\n",
-                get_error_text(error));
-        av_free_packet(&output_packet);
-        return error;
-    }
-
-    /** Write one audio frame from the temporary packet to the output file. */
-    if (*data_present) {
-        if ((error = av_write_frame(output_format_context, &output_packet)) < 0) {
-            fprintf(stderr, "Could not write frame (error '%s')\n",
-                    get_error_text(error));
-            av_free_packet(&output_packet);
-            return error;
-        }
-
-        av_free_packet(&output_packet);
-    }
-
-    return 0;
-}
-
-/**
- * Load one audio frame from the FIFO buffer, encode and write it to the
- * output file.
- */
-static int load_encode_and_write(AVAudioFifo *fifo,
-                                 AVFormatContext *output_format_context,
-                                 AVCodecContext *output_codec_context)
-{
-    /** Temporary storage of the output samples of the frame written to the file. */
-    AVFrame *output_frame;
-    /**
-     * Use the maximum number of possible samples per frame.
-     * If there is less than the maximum possible frame size in the FIFO
-     * buffer use this number. Otherwise, use the maximum possible frame size
-     */
-    const int frame_size = FFMIN(av_audio_fifo_size(fifo),
-                                 output_codec_context->frame_size);
-    int data_written;
-
-    /** Initialize temporary storage for one output frame. */
-    if (init_output_frame(&output_frame, output_codec_context, frame_size))
-        return AVERROR_EXIT;
-
-    /**
-     * Read as many samples from the FIFO buffer as required to fill the frame.
-     * The samples are stored in the frame temporarily.
-     */
-    if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
-        fprintf(stderr, "Could not read data from FIFO\n");
-        av_frame_free(&output_frame);
-        return AVERROR_EXIT;
-    }
-
-    /** Encode one frame worth of audio samples. */
-    if (encode_audio_frame(output_frame, output_format_context,
-                           output_codec_context, &data_written)) {
-        av_frame_free(&output_frame);
-        return AVERROR_EXIT;
-    }
-    av_frame_free(&output_frame);
-    return 0;
-}
-
-/** Write the trailer of the output file container. */
-static int write_output_file_trailer(AVFormatContext *output_format_context)
-{
-    int error;
-    if ((error = av_write_trailer(output_format_context)) < 0) {
-        fprintf(stderr, "Could not write output file trailer (error '%s')\n",
-                get_error_text(error));
-        return error;
-    }
-    return 0;
-}
-
-/** Convert an audio file to an AAC file in an MP4 container. */
-int main(int argc, char **argv)
-{
-    AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
-    AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL;
-    SwrContext *resample_context = NULL;
-    AVAudioFifo *fifo = NULL;
-    int ret = AVERROR_EXIT;
-
-    if (argc < 3) {
-        fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
-        exit(1);
-    }
-
-    /** Register all codecs and formats so that they can be used. */
-    av_register_all();
-    /** Open the input file for reading. */
-    if (open_input_file(argv[1], &input_format_context,
-                        &input_codec_context))
-        goto cleanup;
-    /** Open the output file for writing. */
-    if (open_output_file(argv[2], input_codec_context,
-                         &output_format_context, &output_codec_context))
-        goto cleanup;
-    /** Initialize the resampler to be able to convert audio sample formats. */
-    if (init_resampler(input_codec_context, output_codec_context,
-                       &resample_context))
-        goto cleanup;
-    /** Initialize the FIFO buffer to store audio samples to be encoded. */
-    if (init_fifo(&fifo, output_codec_context))
-        goto cleanup;
-    /** Write the header of the output file container. */
-    if (write_output_file_header(output_format_context))
-        goto cleanup;
-
-    /**
-     * Loop as long as we have input samples to read or output samples
-     * to write; abort as soon as we have neither.
-     */
-    while (1) {
-        /** Use the encoder's desired frame size for processing. */
-        const int output_frame_size = output_codec_context->frame_size;
-        int finished                = 0;
-
-        /**
-         * Make sure that there is one frame worth of samples in the FIFO
-         * buffer so that the encoder can do its work.
-         * Since the decoder's and the encoder's frame size may differ, we
-         * need to FIFO buffer to store as many frames worth of input samples
-         * that they make up at least one frame worth of output samples.
-         */
-        while (av_audio_fifo_size(fifo) < output_frame_size) {
-            /**
-             * Decode one frame worth of audio samples, convert it to the
-             * output sample format and put it into the FIFO buffer.
-             */
-            if (read_decode_convert_and_store(fifo, input_format_context,
-                                              input_codec_context,
-                                              output_codec_context,
-                                              resample_context, &finished))
-                goto cleanup;
-
-            /**
-             * If we are at the end of the input file, we continue
-             * encoding the remaining audio samples to the output file.
-             */
-            if (finished)
-                break;
-        }
-
-        /**
-         * If we have enough samples for the encoder, we encode them.
-         * At the end of the file, we pass the remaining samples to
-         * the encoder.
-         */
-        while (av_audio_fifo_size(fifo) >= output_frame_size ||
-               (finished && av_audio_fifo_size(fifo) > 0))
-            /**
-             * Take one frame worth of audio samples from the FIFO buffer,
-             * encode it and write it to the output file.
-             */
-            if (load_encode_and_write(fifo, output_format_context,
-                                      output_codec_context))
-                goto cleanup;
-
-        /**
-         * If we are at the end of the input file and have encoded
-         * all remaining samples, we can exit this loop and finish.
-         */
-        if (finished) {
-            int data_written;
-            /** Flush the encoder as it may have delayed frames. */
-            do {
-                if (encode_audio_frame(NULL, output_format_context,
-                                       output_codec_context, &data_written))
-                    goto cleanup;
-            } while (data_written);
-            break;
-        }
-    }
-
-    /** Write the trailer of the output file container. */
-    if (write_output_file_trailer(output_format_context))
-        goto cleanup;
-    ret = 0;
-
-cleanup:
-    if (fifo)
-        av_audio_fifo_free(fifo);
-    swr_free(&resample_context);
-    if (output_codec_context)
-        avcodec_close(output_codec_context);
-    if (output_format_context) {
-        avio_closep(&output_format_context->pb);
-        avformat_free_context(output_format_context);
-    }
-    if (input_codec_context)
-        avcodec_close(input_codec_context);
-    if (input_format_context)
-        avformat_close_input(&input_format_context);
-
-    return ret;
-}

doc/examples/transcoding.c

@@ -1,583 +0,0 @@
-/*
- * Copyright (c) 2010 Nicolas George
- * Copyright (c) 2011 Stefano Sabatini
- * Copyright (c) 2014 Andrey Utkin
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be included in
- * all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
- * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- * THE SOFTWARE.
- */
-
-/**
- * @file
- * API example for demuxing, decoding, filtering, encoding and muxing
- * @example transcoding.c
- */
-
-#include <libavcodec/avcodec.h>
-#include <libavformat/avformat.h>
-#include <libavfilter/avfiltergraph.h>
-#include <libavfilter/avcodec.h>
-#include <libavfilter/buffersink.h>
-#include <libavfilter/buffersrc.h>
-#include <libavutil/opt.h>
-#include <libavutil/pixdesc.h>
-
-static AVFormatContext *ifmt_ctx;
-static AVFormatContext *ofmt_ctx;
-typedef struct FilteringContext {
-    AVFilterContext *buffersink_ctx;
-    AVFilterContext *buffersrc_ctx;
-    AVFilterGraph *filter_graph;
-} FilteringContext;
-static FilteringContext *filter_ctx;
-
-static int open_input_file(const char *filename)
-{
-    int ret;
-    unsigned int i;
-
-    ifmt_ctx = NULL;
-    if ((ret = avformat_open_input(&ifmt_ctx, filename, NULL, NULL)) < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
-        return ret;
-    }
-
-    if ((ret = avformat_find_stream_info(ifmt_ctx, NULL)) < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
-        return ret;
-    }
-
-    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
-        AVStream *stream;
-        AVCodecContext *codec_ctx;
-        stream = ifmt_ctx->streams[i];
-        codec_ctx = stream->codec;
-        /* Reencode video & audio and remux subtitles etc. */
-        if (codec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
-                || codec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
-            /* Open decoder */
-            ret = avcodec_open2(codec_ctx,
-                    avcodec_find_decoder(codec_ctx->codec_id), NULL);
-            if (ret < 0) {
-                av_log(NULL, AV_LOG_ERROR, "Failed to open decoder for stream #%u\n", i);
-                return ret;
-            }
-        }
-    }
-
-    av_dump_format(ifmt_ctx, 0, filename, 0);
-    return 0;
-}
-
-static int open_output_file(const char *filename)
-{
-    AVStream *out_stream;
-    AVStream *in_stream;
-    AVCodecContext *dec_ctx, *enc_ctx;
-    AVCodec *encoder;
-    int ret;
-    unsigned int i;
-
-    ofmt_ctx = NULL;
-    avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, filename);
-    if (!ofmt_ctx) {
-        av_log(NULL, AV_LOG_ERROR, "Could not create output context\n");
-        return AVERROR_UNKNOWN;
-    }
-
-
-    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
-        out_stream = avformat_new_stream(ofmt_ctx, NULL);
-        if (!out_stream) {
-            av_log(NULL, AV_LOG_ERROR, "Failed allocating output stream\n");
-            return AVERROR_UNKNOWN;
-        }
-
-        in_stream = ifmt_ctx->streams[i];
-        dec_ctx = in_stream->codec;
-        enc_ctx = out_stream->codec;
-
-        if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
-                || dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
-            /* in this example, we choose transcoding to same codec */
-            encoder = avcodec_find_encoder(dec_ctx->codec_id);
-            if (!encoder) {
-                av_log(NULL, AV_LOG_FATAL, "Neccessary encoder not found\n");
-                return AVERROR_INVALIDDATA;
-            }
-
-            /* In this example, we transcode to same properties (picture size,
-             * sample rate etc.). These properties can be changed for output
-             * streams easily using filters */
-            if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
-                enc_ctx->height = dec_ctx->height;
-                enc_ctx->width = dec_ctx->width;
-                enc_ctx->sample_aspect_ratio = dec_ctx->sample_aspect_ratio;
-                /* take first format from list of supported formats */
-                enc_ctx->pix_fmt = encoder->pix_fmts[0];
-                /* video time_base can be set to whatever is handy and supported by encoder */
-                enc_ctx->time_base = dec_ctx->time_base;
-            } else {
-                enc_ctx->sample_rate = dec_ctx->sample_rate;
-                enc_ctx->channel_layout = dec_ctx->channel_layout;
-                enc_ctx->channels = av_get_channel_layout_nb_channels(enc_ctx->channel_layout);
-                /* take first format from list of supported formats */
-                enc_ctx->sample_fmt = encoder->sample_fmts[0];
-                enc_ctx->time_base = (AVRational){1, enc_ctx->sample_rate};
-            }
-
-            /* Third parameter can be used to pass settings to encoder */
-            ret = avcodec_open2(enc_ctx, encoder, NULL);
-            if (ret < 0) {
-                av_log(NULL, AV_LOG_ERROR, "Cannot open video encoder for stream #%u\n", i);
-                return ret;
-            }
-        } else if (dec_ctx->codec_type == AVMEDIA_TYPE_UNKNOWN) {
-            av_log(NULL, AV_LOG_FATAL, "Elementary stream #%d is of unknown type, cannot proceed\n", i);
-            return AVERROR_INVALIDDATA;
-        } else {
-            /* if this stream must be remuxed */
-            ret = avcodec_copy_context(ofmt_ctx->streams[i]->codec,
-                    ifmt_ctx->streams[i]->codec);
-            if (ret < 0) {
-                av_log(NULL, AV_LOG_ERROR, "Copying stream context failed\n");
-                return ret;
-            }
-        }
-
-        if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
-            enc_ctx->flags |= CODEC_FLAG_GLOBAL_HEADER;
-
-    }
-    av_dump_format(ofmt_ctx, 0, filename, 1);
-
-    if (!(ofmt_ctx->oformat->flags & AVFMT_NOFILE)) {
-        ret = avio_open(&ofmt_ctx->pb, filename, AVIO_FLAG_WRITE);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Could not open output file '%s'", filename);
-            return ret;
-        }
-    }
-
-    /* init muxer, write output file header */
-    ret = avformat_write_header(ofmt_ctx, NULL);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Error occurred when opening output file\n");
-        return ret;
-    }
-
-    return 0;
-}
-
-static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
-        AVCodecContext *enc_ctx, const char *filter_spec)
-{
-    char args[512];
-    int ret = 0;
-    AVFilter *buffersrc = NULL;
-    AVFilter *buffersink = NULL;
-    AVFilterContext *buffersrc_ctx = NULL;
-    AVFilterContext *buffersink_ctx = NULL;
-    AVFilterInOut *outputs = avfilter_inout_alloc();
-    AVFilterInOut *inputs  = avfilter_inout_alloc();
-    AVFilterGraph *filter_graph = avfilter_graph_alloc();
-
-    if (!outputs || !inputs || !filter_graph) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-
-    if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
-        buffersrc = avfilter_get_by_name("buffer");
-        buffersink = avfilter_get_by_name("buffersink");
-        if (!buffersrc || !buffersink) {
-            av_log(NULL, AV_LOG_ERROR, "filtering source or sink element not found\n");
-            ret = AVERROR_UNKNOWN;
-            goto end;
-        }
-
-        snprintf(args, sizeof(args),
-                "video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
-                dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
-                dec_ctx->time_base.num, dec_ctx->time_base.den,
-                dec_ctx->sample_aspect_ratio.num,
-                dec_ctx->sample_aspect_ratio.den);
-
-        ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
-                args, NULL, filter_graph);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
-            goto end;
-        }
-
-        ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
-                NULL, NULL, filter_graph);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
-            goto end;
-        }
-
-        ret = av_opt_set_bin(buffersink_ctx, "pix_fmts",
-                (uint8_t*)&enc_ctx->pix_fmt, sizeof(enc_ctx->pix_fmt),
-                AV_OPT_SEARCH_CHILDREN);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n");
-            goto end;
-        }
-    } else if (dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
-        buffersrc = avfilter_get_by_name("abuffer");
-        buffersink = avfilter_get_by_name("abuffersink");
-        if (!buffersrc || !buffersink) {
-            av_log(NULL, AV_LOG_ERROR, "filtering source or sink element not found\n");
-            ret = AVERROR_UNKNOWN;
-            goto end;
-        }
-
-        if (!dec_ctx->channel_layout)
-            dec_ctx->channel_layout =
-                av_get_default_channel_layout(dec_ctx->channels);
-        snprintf(args, sizeof(args),
-                "time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
-                dec_ctx->time_base.num, dec_ctx->time_base.den, dec_ctx->sample_rate,
-                av_get_sample_fmt_name(dec_ctx->sample_fmt),
-                dec_ctx->channel_layout);
-        ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
-                args, NULL, filter_graph);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
-            goto end;
-        }
-
-        ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
-                NULL, NULL, filter_graph);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
-            goto end;
-        }
-
-        ret = av_opt_set_bin(buffersink_ctx, "sample_fmts",
-                (uint8_t*)&enc_ctx->sample_fmt, sizeof(enc_ctx->sample_fmt),
-                AV_OPT_SEARCH_CHILDREN);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
-            goto end;
-        }
-
-        ret = av_opt_set_bin(buffersink_ctx, "channel_layouts",
-                (uint8_t*)&enc_ctx->channel_layout,
-                sizeof(enc_ctx->channel_layout), AV_OPT_SEARCH_CHILDREN);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
-            goto end;
-        }
-
-        ret = av_opt_set_bin(buffersink_ctx, "sample_rates",
-                (uint8_t*)&enc_ctx->sample_rate, sizeof(enc_ctx->sample_rate),
-                AV_OPT_SEARCH_CHILDREN);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
-            goto end;
-        }
-    } else {
-        ret = AVERROR_UNKNOWN;
-        goto end;
-    }
-
-    /* Endpoints for the filter graph. */
-    outputs->name       = av_strdup("in");
-    outputs->filter_ctx = buffersrc_ctx;
-    outputs->pad_idx    = 0;
-    outputs->next       = NULL;
-
-    inputs->name       = av_strdup("out");
-    inputs->filter_ctx = buffersink_ctx;
-    inputs->pad_idx    = 0;
-    inputs->next       = NULL;
-
-    if (!outputs->name || !inputs->name) {
-        ret = AVERROR(ENOMEM);
-        goto end;
-    }
-
-    if ((ret = avfilter_graph_parse_ptr(filter_graph, filter_spec,
-                    &inputs, &outputs, NULL)) < 0)
-        goto end;
-
-    if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
-        goto end;
-
-    /* Fill FilteringContext */
-    fctx->buffersrc_ctx = buffersrc_ctx;
-    fctx->buffersink_ctx = buffersink_ctx;
-    fctx->filter_graph = filter_graph;
-
-end:
-    avfilter_inout_free(&inputs);
-    avfilter_inout_free(&outputs);
-
-    return ret;
-}
-
-static int init_filters(void)
-{
-    const char *filter_spec;
-    unsigned int i;
-    int ret;
-    filter_ctx = av_malloc_array(ifmt_ctx->nb_streams, sizeof(*filter_ctx));
-    if (!filter_ctx)
-        return AVERROR(ENOMEM);
-
-    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
-        filter_ctx[i].buffersrc_ctx  = NULL;
-        filter_ctx[i].buffersink_ctx = NULL;
-        filter_ctx[i].filter_graph   = NULL;
-        if (!(ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO
-                || ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO))
-            continue;
-
-
-        if (ifmt_ctx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO)
-            filter_spec = "null"; /* passthrough (dummy) filter for video */
-        else
-            filter_spec = "anull"; /* passthrough (dummy) filter for audio */
-        ret = init_filter(&filter_ctx[i], ifmt_ctx->streams[i]->codec,
-                ofmt_ctx->streams[i]->codec, filter_spec);
-        if (ret)
-            return ret;
-    }
-    return 0;
-}
-
-static int encode_write_frame(AVFrame *filt_frame, unsigned int stream_index, int *got_frame) {
-    int ret;
-    int got_frame_local;
-    AVPacket enc_pkt;
-    int (*enc_func)(AVCodecContext *, AVPacket *, const AVFrame *, int *) =
-        (ifmt_ctx->streams[stream_index]->codec->codec_type ==
-         AVMEDIA_TYPE_VIDEO) ? avcodec_encode_video2 : avcodec_encode_audio2;
-
-    if (!got_frame)
-        got_frame = &got_frame_local;
-
-    av_log(NULL, AV_LOG_INFO, "Encoding frame\n");
-    /* encode filtered frame */
-    enc_pkt.data = NULL;
-    enc_pkt.size = 0;
-    av_init_packet(&enc_pkt);
-    ret = enc_func(ofmt_ctx->streams[stream_index]->codec, &enc_pkt,
-            filt_frame, got_frame);
-    av_frame_free(&filt_frame);
-    if (ret < 0)
-        return ret;
-    if (!(*got_frame))
-        return 0;
-
-    /* prepare packet for muxing */
-    enc_pkt.stream_index = stream_index;
-    av_packet_rescale_ts(&enc_pkt,
-                         ofmt_ctx->streams[stream_index]->codec->time_base,
-                         ofmt_ctx->streams[stream_index]->time_base);
-
-    av_log(NULL, AV_LOG_DEBUG, "Muxing frame\n");
-    /* mux encoded frame */
-    ret = av_interleaved_write_frame(ofmt_ctx, &enc_pkt);
-    return ret;
-}
-
-static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
-{
-    int ret;
-    AVFrame *filt_frame;
-
-    av_log(NULL, AV_LOG_INFO, "Pushing decoded frame to filters\n");
-    /* push the decoded frame into the filtergraph */
-    ret = av_buffersrc_add_frame_flags(filter_ctx[stream_index].buffersrc_ctx,
-            frame, 0);
-    if (ret < 0) {
-        av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
-        return ret;
-    }
-
-    /* pull filtered frames from the filtergraph */
-    while (1) {
-        filt_frame = av_frame_alloc();
-        if (!filt_frame) {
-            ret = AVERROR(ENOMEM);
-            break;
-        }
-        av_log(NULL, AV_LOG_INFO, "Pulling filtered frame from filters\n");
-        ret = av_buffersink_get_frame(filter_ctx[stream_index].buffersink_ctx,
-                filt_frame);
-        if (ret < 0) {
-            /* if no more frames for output - returns AVERROR(EAGAIN)
-             * if flushed and no more frames for output - returns AVERROR_EOF
-             * rewrite retcode to 0 to show it as normal procedure completion
-             */
-            if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
-                ret = 0;
-            av_frame_free(&filt_frame);
-            break;
-        }
-
-        filt_frame->pict_type = AV_PICTURE_TYPE_NONE;
-        ret = encode_write_frame(filt_frame, stream_index, NULL);
-        if (ret < 0)
-            break;
-    }
-
-    return ret;
-}
-
-static int flush_encoder(unsigned int stream_index)
-{
-    int ret;
-    int got_frame;
-
-    if (!(ofmt_ctx->streams[stream_index]->codec->codec->capabilities &
-                CODEC_CAP_DELAY))
-        return 0;
-
-    while (1) {
-        av_log(NULL, AV_LOG_INFO, "Flushing stream #%u encoder\n", stream_index);
-        ret = encode_write_frame(NULL, stream_index, &got_frame);
-        if (ret < 0)
-            break;
-        if (!got_frame)
-            return 0;
-    }
-    return ret;
-}
-
-int main(int argc, char **argv)
-{
-    int ret;
-    AVPacket packet = { .data = NULL, .size = 0 };
-    AVFrame *frame = NULL;
-    enum AVMediaType type;
-    unsigned int stream_index;
-    unsigned int i;
-    int got_frame;
-    int (*dec_func)(AVCodecContext *, AVFrame *, int *, const AVPacket *);
-
-    if (argc != 3) {
-        av_log(NULL, AV_LOG_ERROR, "Usage: %s <input file> <output file>\n", argv[0]);
-        return 1;
-    }
-
-    av_register_all();
-    avfilter_register_all();
-
-    if ((ret = open_input_file(argv[1])) < 0)
-        goto end;
-    if ((ret = open_output_file(argv[2])) < 0)
-        goto end;
-    if ((ret = init_filters()) < 0)
-        goto end;
-
-    /* read all packets */
-    while (1) {
-        if ((ret = av_read_frame(ifmt_ctx, &packet)) < 0)
-            break;
-        stream_index = packet.stream_index;
-        type = ifmt_ctx->streams[packet.stream_index]->codec->codec_type;
-        av_log(NULL, AV_LOG_DEBUG, "Demuxer gave frame of stream_index %u\n",
-                stream_index);
-
-        if (filter_ctx[stream_index].filter_graph) {
-            av_log(NULL, AV_LOG_DEBUG, "Going to reencode&filter the frame\n");
-            frame = av_frame_alloc();
-            if (!frame) {
-                ret = AVERROR(ENOMEM);
-                break;
-            }
-            av_packet_rescale_ts(&packet,
-                                 ifmt_ctx->streams[stream_index]->time_base,
-                                 ifmt_ctx->streams[stream_index]->codec->time_base);
-            dec_func = (type == AVMEDIA_TYPE_VIDEO) ? avcodec_decode_video2 :
-                avcodec_decode_audio4;
-            ret = dec_func(ifmt_ctx->streams[stream_index]->codec, frame,
-                    &got_frame, &packet);
-            if (ret < 0) {
-                av_frame_free(&frame);
-                av_log(NULL, AV_LOG_ERROR, "Decoding failed\n");
-                break;
-            }
-
-            if (got_frame) {
-                frame->pts = av_frame_get_best_effort_timestamp(frame);
-                ret = filter_encode_write_frame(frame, stream_index);
-                av_frame_free(&frame);
-                if (ret < 0)
-                    goto end;
-            } else {
-                av_frame_free(&frame);
-            }
-        } else {
-            /* remux this frame without reencoding */
-            av_packet_rescale_ts(&packet,
-                                 ifmt_ctx->streams[stream_index]->time_base,
-                                 ofmt_ctx->streams[stream_index]->time_base);
-
-            ret = av_interleaved_write_frame(ofmt_ctx, &packet);
-            if (ret < 0)
-                goto end;
-        }
-        av_free_packet(&packet);
-    }
-
-    /* flush filters and encoders */
-    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
-        /* flush filter */
-        if (!filter_ctx[i].filter_graph)
-            continue;
-        ret = filter_encode_write_frame(NULL, i);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Flushing filter failed\n");
-            goto end;
-        }
-
-        /* flush encoder */
-        ret = flush_encoder(i);
-        if (ret < 0) {
-            av_log(NULL, AV_LOG_ERROR, "Flushing encoder failed\n");
-            goto end;
-        }
-    }
-
-    av_write_trailer(ofmt_ctx);
-end:
-    av_free_packet(&packet);
-    av_frame_free(&frame);
-    for (i = 0; i < ifmt_ctx->nb_streams; i++) {
-        avcodec_close(ifmt_ctx->streams[i]->codec);
-        if (ofmt_ctx && ofmt_ctx->nb_streams > i && ofmt_ctx->streams[i] && ofmt_ctx->streams[i]->codec)
-            avcodec_close(ofmt_ctx->streams[i]->codec);
-        if (filter_ctx && filter_ctx[i].filter_graph)
-            avfilter_graph_free(&filter_ctx[i].filter_graph);
-    }
-    av_free(filter_ctx);
-    avformat_close_input(&ifmt_ctx);
-    if (ofmt_ctx && !(ofmt_ctx->oformat->flags & AVFMT_NOFILE))
-        avio_closep(&ofmt_ctx->pb);
-    avformat_free_context(ofmt_ctx);
-
-    if (ret < 0)
-        av_log(NULL, AV_LOG_ERROR, "Error occurred: %s\n", av_err2str(ret));
-
-    return ret ? 1 : 0;
-}

doc/faq.texi

@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8
 
 @settitle FFmpeg FAQ
 @titlepage
@@ -91,56 +90,6 @@ To build FFmpeg, you need to install the development package. It is usually
 called @file{libfoo-dev} or @file{libfoo-devel}. You can remove it after the
 build is finished, but be sure to keep the main package.
 
-@section How do I make @command{pkg-config} find my libraries?
-
-Somewhere along with your libraries, there is a @file{.pc} file (or several)
-in a @file{pkgconfig} directory. You need to set environment variables to
-point @command{pkg-config} to these files.
-
-If you need to @emph{add} directories to @command{pkg-config}'s search list
-(typical use case: library installed separately), add it to
-@code{$PKG_CONFIG_PATH}:
-
-@example
-export PKG_CONFIG_PATH=/opt/x264/lib/pkgconfig:/opt/opus/lib/pkgconfig
-@end example
-
-If you need to @emph{replace} @command{pkg-config}'s search list
-(typical use case: cross-compiling), set it in
-@code{$PKG_CONFIG_LIBDIR}:
-
-@example
-export PKG_CONFIG_LIBDIR=/home/me/cross/usr/lib/pkgconfig:/home/me/cross/usr/local/lib/pkgconfig
-@end example
-
-If you need to know the library's internal dependencies (typical use: static
-linking), add the @code{--static} option to @command{pkg-config}:
-
-@example
-./configure --pkg-config-flags=--static
-@end example
-
-@section How do I use @command{pkg-config} when cross-compiling?
-
-The best way is to install @command{pkg-config} in your cross-compilation
-environment. It will automatically use the cross-compilation libraries.
-
-You can also use @command{pkg-config} from the host environment by
-specifying explicitly @code{--pkg-config=pkg-config} to @command{configure}.
-In that case, you must point @command{pkg-config} to the correct directories
-using the @code{PKG_CONFIG_LIBDIR}, as explained in the previous entry.
-
-As an intermediate solution, you can place in your cross-compilation
-environment a script that calls the host @command{pkg-config} with
-@code{PKG_CONFIG_LIBDIR} set. That script can look like that:
-
-@example
-#!/bin/sh
-PKG_CONFIG_LIBDIR=/path/to/cross/lib/pkgconfig
-export PKG_CONFIG_LIBDIR
-exec /usr/bin/pkg-config "$@@"
-@end example
-
 @chapter Usage
 
 @section ffmpeg does not work; what is wrong?
@@ -349,7 +298,7 @@ FFmpeg has a @url{http://ffmpeg.org/ffmpeg-protocols.html#concat,
 @code{concat}} protocol designed specifically for that, with examples in the
 documentation.
 
-A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow one to concatenate
+A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow to concatenate
 video by merely concatenating the files containing them.
 
 Hence you may concatenate your multimedia files by first transcoding them to
@@ -419,6 +368,26 @@ ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \
 rm temp[12].[av] all.[av]
 @end example
 
+@section -profile option fails when encoding H.264 video with AAC audio
+
+@command{ffmpeg} prints an error like
+
+@example
+Undefined constant or missing '(' in 'baseline'
+Unable to parse option value "baseline"
+Error setting option profile to value baseline.
+@end example
+
+Short answer: write @option{-profile:v} instead of @option{-profile}.
+
+Long answer: this happens because the @option{-profile} option can apply to both
+video and audio.  Specifically the AAC encoder also defines some profiles, none
+of which are named @var{baseline}.
+
+The solution is to apply the @option{-profile} option to the video stream only
+by using @url{http://ffmpeg.org/ffmpeg.html#Stream-specifiers-1, Stream specifiers}.
+Appending @code{:v} to it will do exactly that.
+
 @section Using @option{-f lavfi}, audio becomes mono for no apparent reason.
 
 Use @option{-dumpgraph -} to find out exactly where the channel layout is
@@ -443,7 +412,7 @@ VOB and a few other formats do not have a global header that describes
 everything present in the file. Instead, applications are supposed to scan
 the file to see what it contains. Since VOB files are frequently large, only
 the beginning is scanned. If the subtitles happen only later in the file,
-they will not be initially detected.
+they will not be initally detected.
 
 Some applications, including the @code{ffmpeg} command-line tool, can only
 work with streams that were detected during the initial scan; streams that
@@ -467,40 +436,6 @@ point acceptable for your tastes. The most common options to do that are
 @option{-qscale} and @option{-qmax}, but you should peruse the documentation
 of the encoder you chose.
 
-@section I have a stretched video, why does scaling does not fix it?
-
-A lot of video codecs and formats can store the @emph{aspect ratio} of the
-video: this is the ratio between the width and the height of either the full
-image (DAR, display aspect ratio) or individual pixels (SAR, sample aspect
-ratio). For example, EGA screens at resolution 640×350 had 4:3 DAR and 35:48
-SAR.
-
-Most still image processing work with square pixels, i.e. 1:1 SAR, but a lot
-of video standards, especially from the analogic-numeric transition era, use
-non-square pixels.
-
-Most processing filters in FFmpeg handle the aspect ratio to avoid
-stretching the image: cropping adjusts the DAR to keep the SAR constant,
-scaling adjusts the SAR to keep the DAR constant.
-
-If you want to stretch, or “unstretch”, the image, you need to override the
-information with the
-@url{http://ffmpeg.org/ffmpeg-filters.html#setdar_002c-setsar, @code{setdar or setsar filters}}.
-
-Do not forget to examine carefully the original video to check whether the
-stretching comes from the image or from the aspect ratio information.
-
-For example, to fix a badly encoded EGA capture, use the following commands,
-either the first one to upscale to square pixels or the second one to set
-the correct aspect ratio or the third one to avoid transcoding (may not work
-depending on the format / codec / player / phase of the moon):
-
-@example
-ffmpeg -i ega_screen.nut -vf scale=640:480,setsar=1 ega_screen_scaled.nut
-ffmpeg -i ega_screen.nut -vf setdar=4/3 ega_screen_anamorphic.nut
-ffmpeg -i ega_screen.nut -aspect 4/3 -c copy ega_screen_overridden.nut
-@end example
-
 @chapter Development
 
 @section Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat?

doc/fate.texi

@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8
 
 @settitle FFmpeg Automated Testing Environment
 @titlepage

doc/ffmpeg-bitstream-filters.texi

@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8
 
 @settitle FFmpeg Bitstream Filters Documentation
 @titlepage

doc/ffmpeg-codecs.texi

@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8
 
 @settitle FFmpeg Codecs Documentation
 @titlepage

doc/ffmpeg-devices.texi

@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8
 
 @settitle FFmpeg Devices Documentation
 @titlepage

doc/ffmpeg-filters.texi

@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8
 
 @settitle FFmpeg Filters Documentation
 @titlepage

doc/ffmpeg-formats.texi

@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8
 
 @settitle FFmpeg Formats Documentation
 @titlepage

doc/ffmpeg-protocols.texi

@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8
 
 @settitle FFmpeg Protocols Documentation
 @titlepage

doc/ffmpeg-resampler.texi

@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8
 
 @settitle FFmpeg Resampler Documentation
 @titlepage
@@ -15,7 +14,7 @@
 
 The FFmpeg resampler provides a high-level interface to the
 libswresample library audio resampling utilities. In particular it
-allows one to perform audio resampling, audio channel layout rematrixing,
+allows to perform audio resampling, audio channel layout rematrixing,
 and convert audio format and packing layout.
 
 @c man end DESCRIPTION

doc/ffmpeg-scaler.texi

@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8
 
 @settitle FFmpeg Scaler Documentation
 @titlepage
@@ -14,7 +13,7 @@
 @c man begin DESCRIPTION
 
 The FFmpeg rescaler provides a high-level interface to the libswscale
-library image conversion utilities. In particular it allows one to perform
+library image conversion utilities. In particular it allows to perform
 image rescaling and pixel format conversion.
 
 @c man end DESCRIPTION

doc/ffmpeg-utils.texi

@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8
 
 @settitle FFmpeg Utilities Documentation
 @titlepage

doc/ffmpeg.texi

@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8
 
 @settitle ffmpeg Documentation
 @titlepage
@@ -81,23 +80,11 @@ The transcoding process in @command{ffmpeg} for each output can be described by
 the following diagram:
 
 @example
- _______              ______________
-|       |            |              |
-| input |  demuxer   | encoded data |   decoder
-| file  | ---------> | packets      | -----+
-|_______|            |______________|      |
-                                           v
-                                       _________
-                                      |         |
-                                      | decoded |
-                                      | frames  |
-                                      |_________|
- ________             ______________       |
-|        |           |              |      |
-| output | <-------- | encoded data | <----+
-| file   |   muxer   | packets      |   encoder
-|________|           |______________|
-
+ _______              ______________               _________              ______________            ________
+|       |            |              |             |         |            |              |          |        |
+| input |  demuxer   | encoded data |   decoder   | decoded |  encoder   | encoded data |  muxer   | output |
+| file  | ---------> | packets      |  ---------> | frames  | ---------> | packets      | -------> | file   |
+|_______|            |______________|             |_________|            |______________|          |________|
 
 @end example
 
@@ -125,16 +112,11 @@ the same type. In the above diagram they can be represented by simply inserting
 an additional step between decoding and encoding:
 
 @example
- _________                        ______________
-|         |                      |              |
-| decoded |                      | encoded data |
-| frames  |\                   _ | packets      |
-|_________| \                  /||______________|
-             \   __________   /
-  simple     _\||          | /  encoder
-  filtergraph   | filtered |/
-                | frames   |
-                |__________|
+ _________                        __________              ______________
+|         |                      |          |            |              |
+| decoded |  simple filtergraph  | filtered |  encoder   | encoded data |
+| frames  | -------------------> | frames   | ---------> | packets      |
+|_________|                      |__________|            |______________|
 
 @end example
 
@@ -143,10 +125,10 @@ Simple filtergraphs are configured with the per-stream @option{-filter} option
 A simple filtergraph for video can look for example like this:
 
 @example
- _______        _____________        _______        ________
-|       |      |             |      |       |      |        |
-| input | ---> | deinterlace | ---> | scale | ---> | output |
-|_______|      |_____________|      |_______|      |________|
+ _______        _____________        _______        _____        ________
+|       |      |             |      |       |      |     |      |        |
+| input | ---> | deinterlace | ---> | scale | ---> | fps | ---> | output |
+|_______|      |_____________|      |_______|      |_____|      |________|
 
 @end example
 
@@ -273,13 +255,8 @@ ffmpeg -i INPUT -map 0 -c copy -c:v:1 libx264 -c:a:137 libvorbis OUTPUT
 will copy all the streams except the second video, which will be encoded with
 libx264, and the 138th audio, which will be encoded with libvorbis.
 
-@item -t @var{duration} (@emph{input/output})
-When used as an input option (before @code{-i}), limit the @var{duration} of
-data read from the input file.
-
-When used as an output option (before an output filename), stop writing the
-output after its duration reaches @var{duration}.
-
+@item -t @var{duration} (@emph{output})
+Stop writing the output after its duration reaches @var{duration}.
 @var{duration} may be a number in seconds, or in @code{hh:mm:ss[.xxx]} form.
 
 -to and -t are mutually exclusive and -t has priority.
@@ -308,20 +285,23 @@ input until the timestamps reach @var{position}.
 @var{position} may be either in seconds or in @code{hh:mm:ss[.xxx]} form.
 
 @item -itsoffset @var{offset} (@emph{input})
-Set the input time offset.
+Set the input time offset in seconds.
+@code{[-]hh:mm:ss[.xxx]} syntax is also supported.
+The offset is added to the timestamps of the input files.
+Specifying a positive offset means that the corresponding
+streams are delayed by @var{offset} seconds.
 
-@var{offset} must be a time duration specification,
-see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
-
-The offset is added to the timestamps of the input files. Specifying
-a positive offset means that the corresponding streams are delayed by
-the time duration specified in @var{offset}.
-
-@item -timestamp @var{date} (@emph{output})
+@item -timestamp @var{time} (@emph{output})
 Set the recording timestamp in the container.
-
-@var{date} must be a time duration specification,
-see @ref{date syntax,,the Date section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
+The syntax for @var{time} is:
+@example
+now|([(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...])|(HHMMSS[.m...]))[Z|z])
+@end example
+If the value is "now" it takes the current time.
+Time is local time unless 'Z' or 'z' is appended, in which case it is
+interpreted as UTC.
+If the year-month-day part is not specified it takes the current
+year-month-day.
 
 @item -metadata[:metadata_specifier] @var{key}=@var{value} (@emph{output,per-metadata})
 Set a metadata key/value pair.
@@ -340,7 +320,7 @@ ffmpeg -i in.avi -metadata title="my title" out.flv
 
 To set the language of the first audio stream:
 @example
-ffmpeg -i INPUT -metadata:s:a:0 language=eng OUTPUT
+ffmpeg -i INPUT -metadata:s:a:1 language=eng OUTPUT
 @end example
 
 @item -target @var{type} (@emph{output})
@@ -361,20 +341,15 @@ ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg
 @end example
 
 @item -dframes @var{number} (@emph{output})
-Set the number of data frames to output. This is an alias for @code{-frames:d}.
+Set the number of data frames to record. This is an alias for @code{-frames:d}.
 
 @item -frames[:@var{stream_specifier}] @var{framecount} (@emph{output,per-stream})
 Stop writing to the stream after @var{framecount} frames.
 
 @item -q[:@var{stream_specifier}] @var{q} (@emph{output,per-stream})
 @itemx -qscale[:@var{stream_specifier}] @var{q} (@emph{output,per-stream})
-Use fixed quality scale (VBR). The meaning of @var{q}/@var{qscale} is
+Use fixed quality scale (VBR). The meaning of @var{q} is
 codec-dependent.
-If @var{qscale} is used without a @var{stream_specifier} then it applies only
-to the video stream, this is to maintain compatibility with previous behavior
-and as specifying the same codec specific value to 2 different codecs that is
-audio and video generally is not what is intended when no stream_specifier is
-used.
 
 @anchor{filter_option}
 @item -filter[:@var{stream_specifier}] @var{filtergraph} (@emph{output,per-stream})
@@ -468,15 +443,12 @@ attachments.
 
 @table @option
 @item -vframes @var{number} (@emph{output})
-Set the number of video frames to output. This is an alias for @code{-frames:v}.
+Set the number of video frames to record. This is an alias for @code{-frames:v}.
 @item -r[:@var{stream_specifier}] @var{fps} (@emph{input/output,per-stream})
 Set frame rate (Hz value, fraction or abbreviation).
 
 As an input option, ignore any timestamps stored in the file and instead
 generate timestamps assuming constant frame rate @var{fps}.
-This is not the same as the @option{-framerate} option used for some input formats
-like image2 or v4l2 (it used to be the same in older versions of FFmpeg).
-If in doubt use @option{-framerate} instead of the input option @option{-r}.
 
 As an output option, duplicate or drop input frames to achieve constant output
 frame rate @var{fps}.
@@ -531,6 +503,9 @@ prefix is ``ffmpeg2pass''. The complete file name will be
 @file{PREFIX-N.log}, where N is a number specific to the output
 stream
 
+@item -vlang @var{code}
+Set the ISO 639 language code (3 letters) of the current video stream.
+
 @item -vf @var{filtergraph} (@emph{output})
 Create the filtergraph specified by @var{filtergraph} and use it to
 filter the stream.
@@ -538,7 +513,7 @@ filter the stream.
 This is an alias for @code{-filter:v}, see the @ref{filter_option,,-filter option}.
 @end table
 
-@section Advanced Video options
+@section Advanced Video Options
 
 @table @option
 @item -pix_fmt[:@var{stream_specifier}] @var{format} (@emph{input/output,per-stream})
@@ -641,59 +616,13 @@ would be more efficient.
 @item -copyinkf[:@var{stream_specifier}] (@emph{output,per-stream})
 When doing stream copy, copy also non-key frames found at the
 beginning.
-
-@item -hwaccel[:@var{stream_specifier}] @var{hwaccel} (@emph{input,per-stream})
-Use hardware acceleration to decode the matching stream(s). The allowed values
-of @var{hwaccel} are:
-@table @option
-@item none
-Do not use any hardware acceleration (the default).
-
-@item auto
-Automatically select the hardware acceleration method.
-
-@item vda
-Use Apple VDA hardware acceleration.
-
-@item vdpau
-Use VDPAU (Video Decode and Presentation API for Unix) hardware acceleration.
-
-@item dxva2
-Use DXVA2 (DirectX Video Acceleration) hardware acceleration.
-@end table
-
-This option has no effect if the selected hwaccel is not available or not
-supported by the chosen decoder.
-
-Note that most acceleration methods are intended for playback and will not be
-faster than software decoding on modern CPUs. Additionally, @command{ffmpeg}
-will usually need to copy the decoded frames from the GPU memory into the system
-memory, resulting in further performance loss. This option is thus mainly
-useful for testing.
-
-@item -hwaccel_device[:@var{stream_specifier}] @var{hwaccel_device} (@emph{input,per-stream})
-Select a device to use for hardware acceleration.
-
-This option only makes sense when the @option{-hwaccel} option is also
-specified. Its exact meaning depends on the specific hardware acceleration
-method chosen.
-
-@table @option
-@item vdpau
-For VDPAU, this option specifies the X11 display/screen to use. If this option
-is not specified, the value of the @var{DISPLAY} environment variable is used
-
-@item dxva2
-For DXVA2, this option should contain the number of the display adapter to use.
-If this option is not specified, the default adapter is used.
-@end table
 @end table
 
 @section Audio Options
 
 @table @option
 @item -aframes @var{number} (@emph{output})
-Set the number of audio frames to output. This is an alias for @code{-frames:a}.
+Set the number of audio frames to record. This is an alias for @code{-frames:a}.
 @item -ar[:@var{stream_specifier}] @var{freq} (@emph{input/output,per-stream})
 Set the audio sampling frequency. For output streams it is set by
 default to the frequency of the corresponding input stream. For input
@@ -721,7 +650,7 @@ filter the stream.
 This is an alias for @code{-filter:a}, see the @ref{filter_option,,-filter option}.
 @end table
 
-@section Advanced Audio options
+@section Advanced Audio options:
 
 @table @option
 @item -atag @var{fourcc/tag} (@emph{output})
@@ -736,9 +665,11 @@ stereo but not 6 channels as 5.1. The default is to always try to guess. Use
 0 to disable all guessing.
 @end table
 
-@section Subtitle options
+@section Subtitle options:
 
 @table @option
+@item -slang @var{code}
+Set the ISO 639 language code (3 letters) of the current subtitle stream.
 @item -scodec @var{codec} (@emph{input/output})
 Set the subtitle codec. This is an alias for @code{-codec:s}.
 @item -sn (@emph{output})
@@ -747,7 +678,7 @@ Disable subtitle recording.
 Deprecated, see -bsf
 @end table
 
-@section Advanced Subtitle options
+@section Advanced Subtitle options:
 
 @table @option
 
@@ -825,11 +756,6 @@ To map all the streams except the second audio, use negative mappings
 ffmpeg -i INPUT -map 0 -map -0:a:1 OUTPUT
 @end example
 
-To pick the English audio stream:
-@example
-ffmpeg -i INPUT -map 0:m:language:eng OUTPUT
-@end example
-
 Note that using this option disables the default mappings for this output file.
 
 @item -map_channel [@var{input_file_id}.@var{stream_specifier}.@var{channel_id}|-1][:@var{output_file_id}.@var{stream_specifier}]
@@ -995,13 +921,6 @@ With -map you can select from which stream the timestamps should be
 taken. You can leave either video or audio unchanged and sync the
 remaining stream(s) to the unchanged one.
 
-@item -frame_drop_threshold @var{parameter}
-Frame drop threshold, which specifies how much behind video frames can
-be before they are dropped. In frame rate units, so 1.0 is one frame.
-The default is -1.1. One possible usecase is to avoid framedrops in case
-of noisy timestamps or to increase frame drop precision in case of exact
-timestamps.
-
 @item -async @var{samples_per_second}
 Audio sync method. "Stretches/squeezes" the audio stream to match the timestamps,
 the parameter is the maximum samples per second by which the audio is changed.
@@ -1024,12 +943,6 @@ processing (e.g. in case the format option @option{avoid_negative_ts}
 is enabled) the output timestamps may mismatch with the input
 timestamps even when this option is selected.
 
-@item -start_at_zero
-When used with @option{copyts}, shift input timestamps so they start at zero.
-
-This means that using e.g. @code{-ss 50} will make output timestamps start at
-50 seconds, regardless of what timestamp the input file started at.
-
 @item -copytb @var{mode}
 Specify how to set the encoder timebase when stream copying.  @var{mode} is an
 integer numeric value, and can assume one of the following values:
@@ -1085,7 +998,7 @@ ffmpeg -i h264.mp4 -c:v copy -bsf:v h264_mp4toannexb -an out.h264
 ffmpeg -i file.mov -an -vn -bsf:s mov2textsub -c:s copy -f rawvideo sub.txt
 @end example
 
-@item -tag[:@var{stream_specifier}] @var{codec_tag} (@emph{input/output,per-stream})
+@item -tag[:@var{stream_specifier}] @var{codec_tag} (@emph{per-stream})
 Force a tag/fourcc for matching streams.
 
 @item -timecode @var{hh}:@var{mm}:@var{ss}SEP@var{ff}
@@ -1158,50 +1071,13 @@ This option enables or disables accurate seeking in input files with the
 transcoding. Use @option{-noaccurate_seek} to disable it, which may be useful
 e.g. when copying some streams and transcoding the others.
 
-@item -thread_message_queue @var{size} (@emph{input})
-This option sets the maximum number of queued packets when reading from the
-file or device. With low latency / high rate live streams, packets may be
-discarded if they are not read in a timely manner; raising this value can
-avoid it.
-
 @item -override_ffserver (@emph{global})
-Overrides the input specifications from @command{ffserver}. Using this
-option you can map any input stream to @command{ffserver} and control
-many aspects of the encoding from @command{ffmpeg}. Without this
-option @command{ffmpeg} will transmit to @command{ffserver} what is
-requested by @command{ffserver}.
-
+Overrides the input specifications from ffserver. Using this option you can
+map any input stream to ffserver and control many aspects of the encoding from
+ffmpeg. Without this option ffmpeg will transmit to ffserver what is requested by
+ffserver.
 The option is intended for cases where features are needed that cannot be
-specified to @command{ffserver} but can be to @command{ffmpeg}.
-
-@item -sdp_file @var{file} (@emph{global})
-Print sdp information for an output stream to @var{file}.
-This allows dumping sdp information when at least one output isn't an
-rtp stream. (Requires at least one of the output formats to be rtp).
-
-@item -discard (@emph{input})
-Allows discarding specific streams or frames of streams at the demuxer.
-Not all demuxers support this.
-
-@table @option
-@item none
-Discard no frame.
-
-@item default
-Default, which discards no frames.
-
-@item noref
-Discard all non-reference frames.
-
-@item bidir
-Discard all bidirectional frames.
-
-@item nokey
-Discard all frames excepts keyframes.
-
-@item all
-Discard all frames.
-@end table
+specified to ffserver but can be to ffmpeg.
 
 @end table
 
@@ -1228,10 +1104,7 @@ awkward to specify on the command line. Lines starting with the hash
 ('#') character are ignored and are used to provide comments. Check
 the @file{presets} directory in the FFmpeg source tree for examples.
 
-There are two types of preset files: ffpreset and avpreset files.
-
-@subsection ffpreset files
-ffpreset files are specified with the @code{vpre}, @code{apre},
+Preset files are specified with the @code{vpre}, @code{apre},
 @code{spre}, and @code{fpre} options. The @code{fpre} option takes the
 filename of the preset instead of a preset name as input and can be
 used for any kind of codec. For the @code{vpre}, @code{apre}, and
@@ -1256,26 +1129,6 @@ directories, where @var{codec_name} is the name of the codec to which
 the preset file options will be applied. For example, if you select
 the video codec with @code{-vcodec libvpx} and use @code{-vpre 1080p},
 then it will search for the file @file{libvpx-1080p.ffpreset}.
-
-@subsection avpreset files
-avpreset files are specified with the @code{pre} option. They work similar to
-ffpreset files, but they only allow encoder- specific options. Therefore, an
-@var{option}=@var{value} pair specifying an encoder cannot be used.
-
-When the @code{pre} option is specified, ffmpeg will look for files with the
-suffix .avpreset in the directories @file{$AVCONV_DATADIR} (if set), and
-@file{$HOME/.avconv}, and in the datadir defined at configuration time (usually
-@file{PREFIX/share/ffmpeg}), in that order.
-
-First ffmpeg searches for a file named @var{codec_name}-@var{arg}.avpreset in
-the above-mentioned directories, where @var{codec_name} is the name of the codec
-to which the preset file options will be applied. For example, if you select the
-video codec with @code{-vcodec libvpx} and use @code{-pre 1080p}, then it will
-search for the file @file{libvpx-1080p.avpreset}.
-
-If no such file is found, then ffmpeg will search for a file named
-@var{arg}.avpreset in the same directories.
-
 @c man end OPTIONS
 
 @chapter Tips
@@ -1322,6 +1175,21 @@ quality).
 @chapter Examples
 @c man begin EXAMPLES
 
+@section Preset files
+
+A preset file contains a sequence of @var{option=value} pairs, one for
+each line, specifying a sequence of options which can be specified also on
+the command line. Lines starting with the hash ('#') character are ignored and
+are used to provide comments. Empty lines are also ignored. Check the
+@file{presets} directory in the FFmpeg source tree for examples.
+
+Preset files are specified with the @code{pre} option, this option takes a
+preset name as input.  FFmpeg searches for a file named @var{preset_name}.avpreset in
+the directories @file{$AVCONV_DATADIR} (if set), and @file{$HOME/.ffmpeg}, and in
+the data directory defined at configuration time (usually @file{$PREFIX/share/ffmpeg})
+in that order.  For example, if the argument is @code{libx264-max}, it will
+search for the file @file{libx264-max.avpreset}.
+
 @section Video and Audio grabbing
 
 If you specify the input format and device then ffmpeg can grab video
@@ -1491,11 +1359,11 @@ ffmpeg -f image2 -pattern_type glob -i 'foo-*.jpeg' -r 12 -s WxH foo.avi
 You can put many streams of the same type in the output:
 
 @example
-ffmpeg -i test1.avi -i test2.avi -map 1:1 -map 1:0 -map 0:1 -map 0:0 -c copy -y test12.nut
+ffmpeg -i test1.avi -i test2.avi -map 0:3 -map 0:2 -map 0:1 -map 0:0 -c copy test12.nut
 @end example
 
-The resulting output file @file{test12.nut} will contain the first four streams
-from the input files in reverse order.
+The resulting output file @file{test12.avi} will contain first four streams from
+the input file in reverse order.
 
 @item
 To force CBR video output:

doc/ffplay.texi

@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8
 
 @settitle ffplay Documentation
 @titlepage
@@ -38,14 +37,10 @@ Force displayed height.
 Set frame size (WxH or abbreviation), needed for videos which do
 not contain a header with the frame size like raw YUV.  This option
 has been deprecated in favor of private options, try -video_size.
-@item -fs
-Start in fullscreen mode.
 @item -an
 Disable audio.
 @item -vn
 Disable video.
-@item -sn
-Disable subtitles.
 @item -ss @var{pos}
 Seek to a given position in seconds.
 @item -t @var{duration}
@@ -89,9 +84,6 @@ output. In the filtergraph, the input is associated to the label
 ffmpeg-filters manual for more information about the filtergraph
 syntax.
 
-You can specify this parameter multiple times and cycle through the specified
-filtergraphs along with the show modes by pressing the key @key{w}.
-
 @item -af @var{filtergraph}
 @var{filtergraph} is a description of the filtergraph to apply to
 the input audio.
@@ -114,10 +106,15 @@ duration, the codec parameters, the current position in the stream and
 the audio/video synchronisation drift. It is on by default, to
 explicitly disable it you need to specify @code{-nostats}.
 
+@item -bug
+Work around bugs.
 @item -fast
 Non-spec-compliant optimizations.
 @item -genpts
 Generate pts.
+@item -rtp_tcp
+Force RTP/TCP protocol usage instead of RTP/UDP. It is only meaningful
+if you are streaming with the RTSP protocol.
 @item -sync @var{type}
 Set the master clock to audio (@code{type=audio}), video
 (@code{type=video}) or external (@code{type=ext}). Default is audio. The
@@ -125,20 +122,23 @@ master clock is used to control audio-video synchronization. Most media
 players use audio as master clock, but in some cases (streaming or high
 quality broadcast) it is necessary to change that. This option is mainly
 used for debugging purposes.
-@item -ast @var{audio_stream_specifier}
-Select the desired audio stream using the given stream specifier. The stream
-specifiers are described in the @ref{Stream specifiers} chapter. If this option
-is not specified, the "best" audio stream is selected in the program of the
-already selected video stream.
-@item -vst @var{video_stream_specifier}
-Select the desired video stream using the given stream specifier. The stream
-specifiers are described in the @ref{Stream specifiers} chapter. If this option
-is not specified, the "best" video stream is selected.
-@item -sst @var{subtitle_stream_specifier}
-Select the desired subtitle stream using the given stream specifier. The stream
-specifiers are described in the @ref{Stream specifiers} chapter. If this option
-is not specified, the "best" subtitle stream is selected in the program of the
-already selected video or audio stream.
+@item -threads @var{count}
+Set the thread count.
+@item -ast @var{audio_stream_number}
+Select the desired audio stream number, counting from 0. The number
+refers to the list of all the input audio streams. If it is greater
+than the number of audio streams minus one, then the last one is
+selected, if it is negative the audio playback is disabled.
+@item -vst @var{video_stream_number}
+Select the desired video stream number, counting from 0. The number
+refers to the list of all the input video streams. If it is greater
+than the number of video streams minus one, then the last one is
+selected, if it is negative the video playback is disabled.
+@item -sst @var{subtitle_stream_number}
+Select the desired subtitle stream number, counting from 0. The number
+refers to the list of all the input subtitle streams. If it is greater
+than the number of subtitle streams minus one, then the last one is
+selected, if it is negative the subtitle rendering is disabled.
 @item -autoexit
 Exit when video is done playing.
 @item -exitonkeydown
@@ -159,22 +159,6 @@ Force a specific video decoder.
 
 @item -scodec @var{codec_name}
 Force a specific subtitle decoder.
-
-@item -autorotate
-Automatically rotate the video according to presentation metadata. Enabled by
-default, use @option{-noautorotate} to disable it.
-
-@item -framedrop
-Drop video frames if video is out of sync. Enabled by default if the master
-clock is not set to video. Use this option to enable frame dropping for all
-master clock sources, use @option{-noframedrop} to disable it.
-
-@item -infbuf
-Do not limit the input buffer size, read as much data as possible from the
-input as soon as possible. Enabled by default for realtime streams, where data
-may be dropped if not read in time. Use this option to enable infinite buffers
-for all inputs, use @option{-noinfbuf} to disable it.
-
 @end table
 
 @section While playing
@@ -190,7 +174,7 @@ Toggle full screen.
 Pause.
 
 @item a
-Cycle audio channel in the current program.
+Cycle audio channel in the curret program.
 
 @item v
 Cycle video channel.
@@ -202,13 +186,7 @@ Cycle subtitle channel in the current program.
 Cycle program.
 
 @item w
-Cycle video filters or show modes.
-
-@item s
-Step to the next frame.
-
-Pause if the stream is not already paused, step to the next video
-frame, and pause.
+Show audio waves.
 
 @item left/right
 Seek backward/forward 10 seconds.
@@ -217,8 +195,6 @@ Seek backward/forward 10 seconds.
 Seek backward/forward 1 minute.
 
 @item page down/page up
-Seek to the previous/next chapter.
-or if there are no chapters
 Seek backward/forward 10 minutes.
 
 @item mouse click
@@ -230,7 +206,6 @@ Seek to percentage in file corresponding to fraction of width.
 
 @include config.texi
 @ifset config-all
-@set config-readonly
 @ifset config-avutil
 @include utils.texi
 @end ifset

doc/ffprobe.texi

@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8
 
 @settitle ffprobe Documentation
 @titlepage
@@ -120,10 +119,6 @@ Show payload data, as a hexadecimal and ASCII dump. Coupled with
 
 The dump is printed as the "data" field. It may contain newlines.
 
-@item -show_data_hash @var{algorithm}
-Show a hash of payload data, for packets with @option{-show_packets} and for
-codec extradata with @option{-show_streams}.
-
 @item -show_error
 Show information about the error found when trying to probe the input.
 
@@ -185,7 +180,7 @@ format : stream=codec_type
 
 To show all the tags in the stream and format sections:
 @example
-stream_tags : format_tags
+format_tags : format_tags
 @end example
 
 To show only the @code{title} tag (if available) in the stream
@@ -202,11 +197,11 @@ The information for each single packet is printed within a dedicated
 section with name "PACKET".
 
 @item -show_frames
-Show information about each frame and subtitle contained in the input
-multimedia stream.
+Show information about each frame contained in the input multimedia
+stream.
 
 The information for each single frame is printed within a dedicated
-section with name "FRAME" or "SUBTITLE".
+section with name "FRAME".
 
 @item -show_streams
 Show information about each media stream contained in the input
@@ -322,12 +317,6 @@ Show information related to program and library versions. This is the
 equivalent of setting both @option{-show_program_version} and
 @option{-show_library_versions} options.
 
-@item -show_pixel_formats
-Show information about all pixel formats supported by FFmpeg.
-
-Pixel format information for each format is printed within a section
-with name "PIXEL_FORMAT".
-
 @item -bitexact
 Force bitexact output, useful to produce output which is not dependent
 on the specific build.
@@ -348,39 +337,6 @@ A writer may accept one or more arguments, which specify the options
 to adopt. The options are specified as a list of @var{key}=@var{value}
 pairs, separated by ":".
 
-All writers support the following options:
-
-@table @option
-@item string_validation, sv
-Set string validation mode.
-
-The following values are accepted.
-@table @samp
-@item fail
-The writer will fail immediately in case an invalid string (UTF-8)
-sequence or code point is found in the input. This is especially
-useful to validate input metadata.
-
-@item ignore
-Any validation error will be ignored. This will result in possibly
-broken output, especially with the json or xml writer.
-
-@item replace
-The writer will substitute invalid UTF-8 sequences or code points with
-the string specified with the @option{string_validation_replacement}.
-@end table
-
-Default value is @samp{replace}.
-
-@item string_validation_replacement, svr
-Set replacement string to use in case @option{string_validation} is
-set to @samp{replace}.
-
-In case the option is not specified, the writer will assume the empty
-string, that is it will remove the invalid sequences from the input
-strings.
-@end table
-
 A description of the currently available writers follows.
 
 @section default
@@ -610,7 +566,6 @@ DV, GXF and AVI timecodes are available in format metadata
 
 @include config.texi
 @ifset config-all
-@set config-readonly
 @ifset config-avutil
 @include utils.texi
 @end ifset

doc/ffprobe.xsd

@@ -8,17 +8,15 @@
 
     <xsd:complexType name="ffprobeType">
         <xsd:sequence>
-            <xsd:element name="program_version"  type="ffprobe:programVersionType"  minOccurs="0" maxOccurs="1" />
-            <xsd:element name="library_versions" type="ffprobe:libraryVersionsType" minOccurs="0" maxOccurs="1" />
-            <xsd:element name="pixel_formats"    type="ffprobe:pixelFormatsType"    minOccurs="0" maxOccurs="1" />
             <xsd:element name="packets"  type="ffprobe:packetsType" minOccurs="0" maxOccurs="1" />
             <xsd:element name="frames"   type="ffprobe:framesType"  minOccurs="0" maxOccurs="1" />
-            <xsd:element name="packets_and_frames" type="ffprobe:packetsAndFramesType" minOccurs="0" maxOccurs="1" />
-            <xsd:element name="programs" type="ffprobe:programsType" minOccurs="0" maxOccurs="1" />
             <xsd:element name="streams"  type="ffprobe:streamsType" minOccurs="0" maxOccurs="1" />
+            <xsd:element name="programs" type="ffprobe:programsType" minOccurs="0" maxOccurs="1" />
             <xsd:element name="chapters" type="ffprobe:chaptersType" minOccurs="0" maxOccurs="1" />
             <xsd:element name="format"   type="ffprobe:formatType"  minOccurs="0" maxOccurs="1" />
             <xsd:element name="error"    type="ffprobe:errorType"   minOccurs="0" maxOccurs="1" />
+            <xsd:element name="program_version"  type="ffprobe:programVersionType"  minOccurs="0" maxOccurs="1" />
+            <xsd:element name="library_versions" type="ffprobe:libraryVersionsType" minOccurs="0" maxOccurs="1" />
         </xsd:sequence>
     </xsd:complexType>
 
@@ -30,20 +28,7 @@
 
     <xsd:complexType name="framesType">
         <xsd:sequence>
-            <xsd:choice minOccurs="0" maxOccurs="unbounded">
             <xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
-                <xsd:element name="subtitle" type="ffprobe:subtitleType" minOccurs="0" maxOccurs="unbounded"/>
-            </xsd:choice>
-        </xsd:sequence>
-    </xsd:complexType>
-
-    <xsd:complexType name="packetsAndFramesType">
-        <xsd:sequence>
-            <xsd:choice minOccurs="0" maxOccurs="unbounded">
-                <xsd:element name="packet" type="ffprobe:packetType" minOccurs="0" maxOccurs="unbounded"/>
-                <xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
-                <xsd:element name="subtitle" type="ffprobe:subtitleType" minOccurs="0" maxOccurs="unbounded"/>
-            </xsd:choice>
         </xsd:sequence>
     </xsd:complexType>
 
@@ -62,15 +47,9 @@
       <xsd:attribute name="pos"           type="xsd:long"  />
       <xsd:attribute name="flags"         type="xsd:string" use="required" />
       <xsd:attribute name="data"          type="xsd:string" />
-      <xsd:attribute name="data_hash"     type="xsd:string" />
     </xsd:complexType>
 
     <xsd:complexType name="frameType">
-      <xsd:sequence>
-            <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
-            <xsd:element name="side_data_list" type="ffprobe:frameSideDataListType"   minOccurs="0" maxOccurs="1" />
-      </xsd:sequence>
-
       <xsd:attribute name="media_type"    type="xsd:string" use="required"/>
       <xsd:attribute name="key_frame"     type="xsd:int"    use="required"/>
       <xsd:attribute name="pts"           type="xsd:long" />
@@ -79,8 +58,6 @@
       <xsd:attribute name="pkt_pts_time"  type="xsd:float"/>
       <xsd:attribute name="pkt_dts"       type="xsd:long" />
       <xsd:attribute name="pkt_dts_time"  type="xsd:float"/>
-      <xsd:attribute name="best_effort_timestamp"      type="xsd:long" />
-      <xsd:attribute name="best_effort_timestamp_time" type="xsd:float" />
       <xsd:attribute name="pkt_duration"  type="xsd:long" />
       <xsd:attribute name="pkt_duration_time" type="xsd:float"/>
       <xsd:attribute name="pkt_pos"       type="xsd:long" />
@@ -105,26 +82,6 @@
       <xsd:attribute name="repeat_pict"            type="xsd:int"   />
     </xsd:complexType>
 
-    <xsd:complexType name="frameSideDataListType">
-        <xsd:sequence>
-            <xsd:element name="side_data" type="ffprobe:frameSideDataType" minOccurs="1" maxOccurs="unbounded"/>
-        </xsd:sequence>
-    </xsd:complexType>
-    <xsd:complexType name="frameSideDataType">
-        <xsd:attribute name="side_data_type"              type="xsd:string"/>
-        <xsd:attribute name="side_data_size"              type="xsd:int"   />
-    </xsd:complexType>
-
-    <xsd:complexType name="subtitleType">
-      <xsd:attribute name="media_type"         type="xsd:string" fixed="subtitle" use="required"/>
-      <xsd:attribute name="pts"                type="xsd:long" />
-      <xsd:attribute name="pts_time"           type="xsd:float"/>
-      <xsd:attribute name="format"             type="xsd:int"  />
-      <xsd:attribute name="start_display_time" type="xsd:int"  />
-      <xsd:attribute name="end_display_time"   type="xsd:int"  />
-      <xsd:attribute name="num_rects"          type="xsd:int"  />
-    </xsd:complexType>
-
     <xsd:complexType name="streamsType">
         <xsd:sequence>
             <xsd:element name="stream" type="ffprobe:streamType" minOccurs="0" maxOccurs="unbounded"/>
@@ -166,7 +123,6 @@
       <xsd:attribute name="codec_tag"        type="xsd:string" use="required"/>
       <xsd:attribute name="codec_tag_string" type="xsd:string" use="required"/>
       <xsd:attribute name="extradata"        type="xsd:string" />
-      <xsd:attribute name="extradata_hash"   type="xsd:string" />
 
       <!-- video attributes -->
       <xsd:attribute name="width"                type="xsd:int"/>
@@ -176,13 +132,7 @@
       <xsd:attribute name="display_aspect_ratio" type="xsd:string"/>
       <xsd:attribute name="pix_fmt"              type="xsd:string"/>
       <xsd:attribute name="level"                type="xsd:int"/>
-      <xsd:attribute name="color_range"          type="xsd:string"/>
-      <xsd:attribute name="color_space"          type="xsd:string"/>
-      <xsd:attribute name="color_transfer"       type="xsd:string"/>
-      <xsd:attribute name="color_primaries"      type="xsd:string"/>
-      <xsd:attribute name="chroma_location"      type="xsd:string"/>
       <xsd:attribute name="timecode"             type="xsd:string"/>
-      <xsd:attribute name="refs"                 type="xsd:int"/>
 
       <!-- audio attributes -->
       <xsd:attribute name="sample_fmt"       type="xsd:string"/>
@@ -200,8 +150,6 @@
       <xsd:attribute name="duration_ts"      type="xsd:long"/>
       <xsd:attribute name="duration"         type="xsd:float"/>
       <xsd:attribute name="bit_rate"         type="xsd:int"/>
-      <xsd:attribute name="max_bit_rate"     type="xsd:int"/>
-      <xsd:attribute name="bits_per_raw_sample" type="xsd:int"/>
       <xsd:attribute name="nb_frames"        type="xsd:int"/>
       <xsd:attribute name="nb_read_frames"   type="xsd:int"/>
       <xsd:attribute name="nb_read_packets"  type="xsd:int"/>
@@ -254,7 +202,10 @@
     <xsd:complexType name="programVersionType">
       <xsd:attribute name="version"          type="xsd:string" use="required"/>
       <xsd:attribute name="copyright"        type="xsd:string" use="required"/>
-      <xsd:attribute name="compiler_ident"   type="xsd:string" use="required"/>
+      <xsd:attribute name="build_date"       type="xsd:string" use="required"/>
+      <xsd:attribute name="build_time"       type="xsd:string" use="required"/>
+      <xsd:attribute name="compiler_type"    type="xsd:string" use="required"/>
+      <xsd:attribute name="compiler_version" type="xsd:string" use="required"/>
       <xsd:attribute name="configuration"    type="xsd:string" use="required"/>
     </xsd:complexType>
 
@@ -291,45 +242,4 @@
           <xsd:element name="library_version" type="ffprobe:libraryVersionType" minOccurs="0" maxOccurs="unbounded"/>
         </xsd:sequence>
     </xsd:complexType>
-
-    <xsd:complexType name="pixelFormatFlagsType">
-      <xsd:attribute name="big_endian" type="xsd:int" use="required"/>
-      <xsd:attribute name="palette"    type="xsd:int" use="required"/>
-      <xsd:attribute name="bitstream"  type="xsd:int" use="required"/>
-      <xsd:attribute name="hwaccel"    type="xsd:int" use="required"/>
-      <xsd:attribute name="planar"     type="xsd:int" use="required"/>
-      <xsd:attribute name="rgb"        type="xsd:int" use="required"/>
-      <xsd:attribute name="pseudopal"  type="xsd:int" use="required"/>
-      <xsd:attribute name="alpha"      type="xsd:int" use="required"/>
-    </xsd:complexType>
-
-    <xsd:complexType name="pixelFormatComponentType">
-      <xsd:attribute name="index"      type="xsd:int" use="required"/>
-      <xsd:attribute name="bit_depth"  type="xsd:int" use="required"/>
-    </xsd:complexType>
-
-    <xsd:complexType name="pixelFormatComponentsType">
-      <xsd:sequence>
-        <xsd:element name="component" type="ffprobe:pixelFormatComponentType" minOccurs="0" maxOccurs="unbounded"/>
-      </xsd:sequence>
-    </xsd:complexType>
-
-    <xsd:complexType name="pixelFormatType">
-      <xsd:sequence>
-        <xsd:element name="flags" type="ffprobe:pixelFormatFlagsType" minOccurs="0" maxOccurs="1"/>
-        <xsd:element name="components" type="ffprobe:pixelFormatComponentsType" minOccurs="0" maxOccurs="1"/>
-      </xsd:sequence>
-
-      <xsd:attribute name="name"                type="xsd:string" use="required"/>
-      <xsd:attribute name="nb_components"       type="xsd:int"    use="required"/>
-      <xsd:attribute name="log2_chroma_w"       type="xsd:int"/>
-      <xsd:attribute name="log2_chroma_h"       type="xsd:int"/>
-      <xsd:attribute name="bits_per_pixel"      type="xsd:int"/>
-    </xsd:complexType>
-
-    <xsd:complexType name="pixelFormatsType">
-      <xsd:sequence>
-        <xsd:element name="pixel_format" type="ffprobe:pixelFormatType" minOccurs="0" maxOccurs="unbounded"/>
-      </xsd:sequence>
-    </xsd:complexType>
 </xsd:schema>

doc/ffserver.conf

@@ -1,11 +1,11 @@
 # Port on which the server is listening. You must select a different
 # port from your standard HTTP web server if it is running on the same
 # computer.
-HTTPPort 8090
+Port 8090
 
 # Address on which the server is bound. Only useful if you have
 # several network interfaces.
-HTTPBindAddress 0.0.0.0
+BindAddress 0.0.0.0
 
 # Number of simultaneous HTTP connections that can be handled. It has
 # to be defined *before* the MaxClients parameter, since it defines the
@@ -235,7 +235,7 @@ StartSendOnKey
 
 #<Stream test.ogg>
 #Feed feed1.ffm
-#Metadata title "Stream title"
+#Title "Stream title"
 #AudioBitRate 64
 #AudioChannels 2
 #AudioSampleRate 44100
@@ -280,10 +280,10 @@ StartSendOnKey
 #<Stream file.asf>
 #File "/usr/local/httpd/htdocs/test.asf"
 #NoAudio
-#Metadata author "Me"
-#Metadata copyright "Super MegaCorp"
-#Metadata title "Test stream from disk"
-#Metadata comment "Test comment"
+#Author "Me"
+#Copyright "Super MegaCorp"
+#Title "Test stream from disk"
+#Comment "Test comment"
 #</Stream>
 
 

doc/ffserver.texi

@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8
 
 @settitle ffserver Documentation
 @titlepage
@@ -17,14 +16,11 @@ ffserver [@var{options}]
 @chapter Description
 @c man begin DESCRIPTION
 
-@command{ffserver} is a streaming server for both audio and video.
-It supports several live feeds, streaming from files and time shifting
-on live feeds. You can seek to positions in the past on each live
-feed, provided you specify a big enough feed storage.
-
-@command{ffserver} is configured through a configuration file, which
-is read at startup. If not explicitly specified, it will read from
-@file{/etc/ffserver.conf}.
+@command{ffserver} is a streaming server for both audio and video. It
+supports several live feeds, streaming from files and time shifting on
+live feeds (you can seek to positions in the past on each live feed,
+provided you specify a big enough feed storage in
+@file{ffserver.conf}).
 
 @command{ffserver} receives prerecorded files or FFM streams from some
 @command{ffmpeg} instance as input, then streams them over
@@ -43,126 +39,10 @@ For each feed you can have different output streams in various
 formats, each one specified by a @code{<Stream>} section in the
 configuration file.
 
-@chapter Detailed description
-
-@command{ffserver} works by forwarding streams encoded by
-@command{ffmpeg}, or pre-recorded streams which are read from disk.
-
-Precisely, @command{ffserver} acts as an HTTP server, accepting POST
-requests from @command{ffmpeg} to acquire the stream to publish, and
-serving RTSP clients or HTTP clients GET requests with the stream
-media content.
-
-A feed is an @ref{FFM} stream created by @command{ffmpeg}, and sent to
-a port where @command{ffserver} is listening.
-
-Each feed is identified by a unique name, corresponding to the name
-of the resource published on @command{ffserver}, and is configured by
-a dedicated @code{Feed} section in the configuration file.
-
-The feed publish URL is given by:
-@example
-http://@var{ffserver_ip_address}:@var{http_port}/@var{feed_name}
-@end example
-
-where @var{ffserver_ip_address} is the IP address of the machine where
-@command{ffserver} is installed, @var{http_port} is the port number of
-the HTTP server (configured through the @option{HTTPPort} option), and
-@var{feed_name} is the name of the corresponding feed defined in the
-configuration file.
-
-Each feed is associated to a file which is stored on disk. This stored
-file is used to send pre-recorded data to a player as fast as
-possible when new content is added in real-time to the stream.
-
-A "live-stream" or "stream" is a resource published by
-@command{ffserver}, and made accessible through the HTTP protocol to
-clients.
-
-A stream can be connected to a feed, or to a file. In the first case,
-the published stream is forwarded from the corresponding feed
-generated by a running instance of @command{ffmpeg}, in the second
-case the stream is read from a pre-recorded file.
-
-Each stream is identified by a unique name, corresponding to the name
-of the resource served by @command{ffserver}, and is configured by
-a dedicated @code{Stream} section in the configuration file.
-
-The stream access HTTP URL is given by:
-@example
-http://@var{ffserver_ip_address}:@var{http_port}/@var{stream_name}[@var{options}]
-@end example
-
-The stream access RTSP URL is given by:
-@example
-http://@var{ffserver_ip_address}:@var{rtsp_port}/@var{stream_name}[@var{options}]
-@end example
-
-@var{stream_name} is the name of the corresponding stream defined in
-the configuration file. @var{options} is a list of options specified
-after the URL which affects how the stream is served by
-@command{ffserver}. @var{http_port} and @var{rtsp_port} are the HTTP
-and RTSP ports configured with the options @var{HTTPPort} and
-@var{RTSPPort} respectively.
-
-In case the stream is associated to a feed, the encoding parameters
-must be configured in the stream configuration. They are sent to
-@command{ffmpeg} when setting up the encoding. This allows
-@command{ffserver} to define the encoding parameters used by
-the @command{ffmpeg} encoders.
-
-The @command{ffmpeg} @option{override_ffserver} commandline option
-allows one to override the encoding parameters set by the server.
-
-Multiple streams can be connected to the same feed.
-
-For example, you can have a situation described by the following
-graph:
-@example
-               _________       __________
-              |         |     |          |
-ffmpeg 1 -----| feed 1  |-----| stream 1 |
-    \         |_________|\    |__________|
-     \                    \
-      \                    \   __________
-       \                    \ |          |
-        \                    \| stream 2 |
-         \                    |__________|
-          \
-           \   _________       __________
-            \ |         |     |          |
-             \| feed 2  |-----| stream 3 |
-              |_________|     |__________|
-
-               _________       __________
-              |         |     |          |
-ffmpeg 2 -----| feed 3  |-----| stream 4 |
-              |_________|     |__________|
-
-               _________       __________
-              |         |     |          |
-              | file 1  |-----| stream 5 |
-              |_________|     |__________|
-@end example
-
-@anchor{FFM}
-@section FFM, FFM2 formats
-
-FFM and FFM2 are formats used by ffserver. They allow storing a wide variety of
-video and audio streams and encoding options, and can store a moving time segment
-of an infinite movie or a whole movie.
-
-FFM is version specific, and there is limited compatibility of FFM files
-generated by one version of ffmpeg/ffserver and another version of
-ffmpeg/ffserver. It may work but it is not guaranteed to work.
-
-FFM2 is extensible while maintaining compatibility and should work between
-differing versions of tools. FFM2 is the default.
-
 @section Status stream
 
-@command{ffserver} supports an HTTP interface which exposes the
-current status of the server.
+ffserver supports an HTTP interface which exposes the current status
+of the server.
 
 Simply point your browser to the address of the special status stream
 specified in the configuration file.
@@ -181,8 +61,27 @@ ACL allow 192.168.0.0 192.168.255.255
 then the server will post a page with the status information when
 the special stream @file{status.html} is requested.
 
+@section What can this do?
+
+When properly configured and running, you can capture video and audio in real
+time from a suitable capture card, and stream it out over the Internet to
+either Windows Media Player or RealAudio player (with some restrictions).
+
+It can also stream from files, though that is currently broken. Very often, a
+web server can be used to serve up the files just as well.
+
+It can stream prerecorded video from .ffm files, though it is somewhat tricky
+to make it work correctly.
+
 @section How do I make it work?
 
+First, build the kit. It *really* helps to have installed LAME first. Then when
+you run the ffserver ./configure, make sure that you have the
+@code{--enable-libmp3lame} flag turned on.
+
+LAME is important as it allows for streaming audio to Windows Media Player.
+Don't ask why the other audio types do not work.
+
 As a simple test, just run the following two command lines where INPUTFILE
 is some file which you can decode with ffmpeg:
 
@@ -204,9 +103,40 @@ WARNING: trying to stream test1.mpg doesn't work with WMP as it tries to
 transfer the entire file before starting to play.
 The same is true of AVI files.
 
-You should edit the @file{ffserver.conf} file to suit your needs (in
-terms of frame rates etc). Then install @command{ffserver} and
-@command{ffmpeg}, write a script to start them up, and off you go.
+@section What happens next?
+
+You should edit the ffserver.conf file to suit your needs (in terms of
+frame rates etc). Then install ffserver and ffmpeg, write a script to start
+them up, and off you go.
+
+@section Troubleshooting
+
+@subsection I don't hear any audio, but video is fine.
+
+Maybe you didn't install LAME, or got your ./configure statement wrong. Check
+the ffmpeg output to see if a line referring to MP3 is present. If not, then
+your configuration was incorrect. If it is, then maybe your wiring is not
+set up correctly. Maybe the sound card is not getting data from the right
+input source. Maybe you have a really awful audio interface (like I do)
+that only captures in stereo and also requires that one channel be flipped.
+If you are one of these people, then export 'AUDIO_FLIP_LEFT=1' before
+starting ffmpeg.
+
+@subsection The audio and video lose sync after a while.
+
+Yes, they do.
+
+@subsection After a long while, the video update rate goes way down in WMP.
+
+Yes, it does. Who knows why?
+
+@subsection WMP 6.4 behaves differently to WMP 7.
+
+Yes, it does. Any thoughts on this would be gratefully received. These
+differences extend to embedding WMP into a web page. [There are two
+object IDs that you can use: The old one, which does not play well, and
+the new one, which does (both tested on the same system). However,
+I suspect that the new one is not available unless you have installed WMP 7].
 
 @section What else can it do?
 
@@ -247,6 +177,9 @@ specify a time. In addition, ffserver will skip frames until a key_frame
 is found. This further reduces the startup delay by not transferring data
 that will be discarded.
 
+* You may want to adjust the MaxBandwidth in the ffserver.conf to limit
+the amount of bandwidth consumed by live streams.
+
 @section Why does the ?buffer / Preroll stop working after a time?
 
 It turns out that (on my machine at least) the number of frames successfully
@@ -280,6 +213,19 @@ You use this by adding the ?date= to the end of the URL for the stream.
 For example:   @samp{http://localhost:8080/test.asf?date=2002-07-26T23:05:00}.
 @c man end
 
+@section What is FFM, FFM2
+
+FFM and FFM2 are formats used by ffserver. They allow storing a wide variety of
+video and audio streams and encoding options, and can store a moving time segment
+of an infinite movie or a whole movie.
+
+FFM is version specific, and there is limited compatibility of FFM files
+generated by one version of ffmpeg/ffserver and another version of
+ffmpeg/ffserver. It may work but it is not guaranteed to work.
+
+FFM2 is extensible while maintaining compatibility and should work between
+differing versions of tools. FFM2 is the default.
+
 @chapter Options
 @c man begin OPTIONS
 
@@ -289,562 +235,15 @@ For example:   @samp{http://localhost:8080/test.asf?date=2002-07-26T23:05:00}.
 
 @table @option
 @item -f @var{configfile}
-Read configuration file @file{configfile}. If not specified it will
-read by default from @file{/etc/ffserver.conf}.
-
+Use @file{configfile} instead of @file{/etc/ffserver.conf}.
 @item -n
-Enable no-launch mode. This option disables all the @code{Launch}
-directives within the various @code{<Feed>} sections. Since
-@command{ffserver} will not launch any @command{ffmpeg} instances, you
-will have to launch them manually.
-
+Enable no-launch mode. This option disables all the Launch directives
+within the various <Stream> sections. Since ffserver will not launch
+any ffmpeg instances, you will have to launch them manually.
 @item -d
-Enable debug mode. This option increases log verbosity, and directs
-log messages to stdout. When specified, the @option{CustomLog} option
-is ignored.
+Enable debug mode. This option increases log verbosity, directs log
+messages to stdout.
 @end table
-
-@chapter Configuration file syntax
-
-@command{ffserver} reads a configuration file containing global
-options and settings for each stream and feed.
-
-The configuration file consists of global options and dedicated
-sections, which must be introduced by "<@var{SECTION_NAME}
-@var{ARGS}>" on a separate line and must be terminated by a line in
-the form "</@var{SECTION_NAME}>". @var{ARGS} is optional.
-
-Currently the following sections are recognized: @samp{Feed},
-@samp{Stream}, @samp{Redirect}.
-
-A line starting with @code{#} is ignored and treated as a comment.
-
-Name of options and sections are case-insensitive.
-
-@section ACL syntax
-An ACL (Access Control List) specifies the address which are allowed
-to access a given stream, or to write a given feed.
-
-It accepts the folling forms
-@itemize
-@item
-Allow/deny access to @var{address}.
-@example
-ACL ALLOW <address>
-ACL DENY <address>
-@end example
-
-@item
-Allow/deny access to ranges of addresses from @var{first_address} to
-@var{last_address}.
-@example
-ACL ALLOW <first_address> <last_address>
-ACL DENY <first_address> <last_address>
-@end example
-@end itemize
-
-You can repeat the ACL allow/deny as often as you like. It is on a per
-stream basis. The first match defines the action. If there are no matches,
-then the default is the inverse of the last ACL statement.
-
-Thus 'ACL allow localhost' only allows access from localhost.
-'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and
-allow everybody else.
-
-@section Global options
-@table @option
-@item HTTPPort @var{port_number}
-@item Port @var{port_number}
-@item RTSPPort @var{port_number}
-
-@var{HTTPPort} sets the HTTP server listening TCP port number,
-@var{RTSPPort} sets the RTSP server listening TCP port number.
-
-@var{Port} is the equivalent of @var{HTTPPort} and is deprecated.
-
-You must select a different port from your standard HTTP web server if
-it is running on the same computer.
-
-If not specified, no corresponding server will be created.
-
-@item HTTPBindAddress @var{ip_address}
-@item BindAddress @var{ip_address}
-@item RTSPBindAddress @var{ip_address}
-Set address on which the HTTP/RTSP server is bound. Only useful if you
-have several network interfaces.
-
-@var{BindAddress} is the equivalent of @var{HTTPBindAddress} and is
-deprecated.
-
-@item MaxHTTPConnections @var{n}
-Set number of simultaneous HTTP connections that can be handled. It
-has to be defined @emph{before} the @option{MaxClients} parameter,
-since it defines the @option{MaxClients} maximum limit.
-
-Default value is 2000.
-
-@item MaxClients @var{n}
-Set number of simultaneous requests that can be handled. Since
-@command{ffserver} is very fast, it is more likely that you will want
-to leave this high and use @option{MaxBandwidth}.
-
-Default value is 5.
-
-@item MaxBandwidth @var{kbps}
-Set the maximum amount of kbit/sec that you are prepared to consume
-when streaming to clients.
-
-Default value is 1000.
-
-@item CustomLog @var{filename}
-Set access log file (uses standard Apache log file format). '-' is the
-standard output.
-
-If not specified @command{ffserver} will produce no log.
-
-In case the commandline option @option{-d} is specified this option is
-ignored, and the log is written to standard output.
-
-@item NoDaemon
-Set no-daemon mode. This option is currently ignored since now
-@command{ffserver} will always work in no-daemon mode, and is
-deprecated.
-
-@item UseDefaults
-@item NoDefaults
-Control whether default codec options are used for the all streams or not.
-Each stream may overwrite this setting for its own. Default is @var{UseDefaults}.
-The lastest occurrence overrides previous if multiple definitions.
-@end table
-
-@section Feed section
-
-A Feed section defines a feed provided to @command{ffserver}.
-
-Each live feed contains one video and/or audio sequence coming from an
-@command{ffmpeg} encoder or another @command{ffserver}. This sequence
-may be encoded simultaneously with several codecs at several
-resolutions.
-
-A feed instance specification is introduced by a line in the form:
-@example
-<Feed FEED_FILENAME>
-@end example
-
-where @var{FEED_FILENAME} specifies the unique name of the FFM stream.
-
-The following options are recognized within a Feed section.
-
-@table @option
-@item File @var{filename}
-@item ReadOnlyFile @var{filename}
-Set the path where the feed file is stored on disk.
-
-If not specified, the @file{/tmp/FEED.ffm} is assumed, where
-@var{FEED} is the feed name.
-
-If @option{ReadOnlyFile} is used the file is marked as read-only and
-it will not be deleted or updated.
-
-@item Truncate
-Truncate the feed file, rather than appending to it. By default
-@command{ffserver} will append data to the file, until the maximum
-file size value is reached (see @option{FileMaxSize} option).
-
-@item FileMaxSize @var{size}
-Set maximum size of the feed file in bytes. 0 means unlimited. The
-postfixes @code{K} (2^10), @code{M} (2^20), and @code{G} (2^30) are
-recognized.
-
-Default value is 5M.
-
-@item Launch @var{args}
-Launch an @command{ffmpeg} command when creating @command{ffserver}.
-
-@var{args} must be a sequence of arguments to be provided to an
-@command{ffmpeg} instance. The first provided argument is ignored, and
-it is replaced by a path with the same dirname of the @command{ffserver}
-instance, followed by the remaining argument and terminated with a
-path corresponding to the feed.
-
-When the launched process exits, @command{ffserver} will launch
-another program instance.
-
-In case you need a more complex @command{ffmpeg} configuration,
-e.g. if you need to generate multiple FFM feeds with a single
-@command{ffmpeg} instance, you should launch @command{ffmpeg} by hand.
-
-This option is ignored in case the commandline option @option{-n} is
-specified.
-
-@item ACL @var{spec}
-Specify the list of IP address which are allowed or denied to write
-the feed. Multiple ACL options can be specified.
-@end table
-
-@section Stream section
-
-A Stream section defines a stream provided by @command{ffserver}, and
-identified by a single name.
-
-The stream is sent when answering a request containing the stream
-name.
-
-A stream section must be introduced by the line:
-@example
-<Stream STREAM_NAME>
-@end example
-
-where @var{STREAM_NAME} specifies the unique name of the stream.
-
-The following options are recognized within a Stream section.
-
-Encoding options are marked with the @emph{encoding} tag, and they are
-used to set the encoding parameters, and are mapped to libavcodec
-encoding options. Not all encoding options are supported, in
-particular it is not possible to set encoder private options. In order
-to override the encoding options specified by @command{ffserver}, you
-can use the @command{ffmpeg} @option{override_ffserver} commandline
-option.
-
-Only one of the @option{Feed} and @option{File} options should be set.
-
-@table @option
-@item Feed @var{feed_name}
-Set the input feed. @var{feed_name} must correspond to an existing
-feed defined in a @code{Feed} section.
-
-When this option is set, encoding options are used to setup the
-encoding operated by the remote @command{ffmpeg} process.
-
-@item File @var{filename}
-Set the filename of the pre-recorded input file to stream.
-
-When this option is set, encoding options are ignored and the input
-file content is re-streamed as is.
-
-@item Format @var{format_name}
-Set the format of the output stream.
-
-Must be the name of a format recognized by FFmpeg. If set to
-@samp{status}, it is treated as a status stream.
-
-@item InputFormat @var{format_name}
-Set input format. If not specified, it is automatically guessed.
-
-@item Preroll @var{n}
-Set this to the number of seconds backwards in time to start. Note that
-most players will buffer 5-10 seconds of video, and also you need to allow
-for a keyframe to appear in the data stream.
-
-Default value is 0.
-
-@item StartSendOnKey
-Do not send stream until it gets the first key frame. By default
-@command{ffserver} will send data immediately.
-
-@item MaxTime @var{n}
-Set the number of seconds to run. This value set the maximum duration
-of the stream a client will be able to receive.
-
-A value of 0 means that no limit is set on the stream duration.
-
-@item ACL @var{spec}
-Set ACL for the stream.
-
-@item DynamicACL @var{spec}
-
-@item RTSPOption @var{option}
-
-@item MulticastAddress @var{address}
-
-@item MulticastPort @var{port}
-
-@item MulticastTTL @var{integer}
-
-@item NoLoop
-
-@item FaviconURL @var{url}
-Set favicon (favourite icon) for the server status page. It is ignored
-for regular streams.
-
-@item Author @var{value}
-@item Comment @var{value}
-@item Copyright @var{value}
-@item Title @var{value}
-Set metadata corresponding to the option. All these options are
-deprecated in favor of @option{Metadata}.
-
-@item Metadata @var{key} @var{value}
-Set metadata value on the output stream.
-
-@item UseDefaults
-@item NoDefaults
-Control whether default codec options are used for the stream or not.
-Default is @var{UseDefaults} unless disabled globally.
-
-@item NoAudio
-@item NoVideo
-Suppress audio/video.
-
-@item AudioCodec @var{codec_name} (@emph{encoding,audio})
-Set audio codec.
-
-@item AudioBitRate @var{rate} (@emph{encoding,audio})
-Set bitrate for the audio stream in kbits per second.
-
-@item AudioChannels @var{n} (@emph{encoding,audio})
-Set number of audio channels.
-
-@item AudioSampleRate @var{n} (@emph{encoding,audio})
-Set sampling frequency for audio. When using low bitrates, you should
-lower this frequency to 22050 or 11025. The supported frequencies
-depend on the selected audio codec.
-
-@item AVOptionAudio [@var{codec}:]@var{option} @var{value} (@emph{encoding,audio})
-Set generic or private option for audio stream.
-Private option must be prefixed with codec name or codec must be defined before.
-
-@item AVPresetAudio @var{preset} (@emph{encoding,audio})
-Set preset for audio stream.
-
-@item VideoCodec @var{codec_name} (@emph{encoding,video})
-Set video codec.
-
-@item VideoBitRate @var{n} (@emph{encoding,video})
-Set bitrate for the video stream in kbits per second.
-
-@item VideoBitRateRange @var{range} (@emph{encoding,video})
-Set video bitrate range.
-
-A range must be specified in the form @var{minrate}-@var{maxrate}, and
-specifies the @option{minrate} and @option{maxrate} encoding options
-expressed in kbits per second.
-
-@item VideoBitRateRangeTolerance @var{n} (@emph{encoding,video})
-Set video bitrate tolerance in kbits per second.
-
-@item PixelFormat @var{pixel_format} (@emph{encoding,video})
-Set video pixel format.
-
-@item Debug @var{integer} (@emph{encoding,video})
-Set video @option{debug} encoding option.
-
-@item Strict @var{integer} (@emph{encoding,video})
-Set video @option{strict} encoding option.
-
-@item VideoBufferSize @var{n} (@emph{encoding,video})
-Set ratecontrol buffer size, expressed in KB.
-
-@item VideoFrameRate @var{n} (@emph{encoding,video})
-Set number of video frames per second.
-
-@item VideoSize (@emph{encoding,video})
-Set size of the video frame, must be an abbreviation or in the form
-@var{W}x@var{H}.  See @ref{video size syntax,,the Video size section
-in the ffmpeg-utils(1) manual,ffmpeg-utils}.
-
-Default value is @code{160x128}.
-
-@item VideoIntraOnly (@emph{encoding,video})
-Transmit only intra frames (useful for low bitrates, but kills frame rate).
-
-@item VideoGopSize @var{n} (@emph{encoding,video})
-If non-intra only, an intra frame is transmitted every VideoGopSize
-frames. Video synchronization can only begin at an intra frame.
-
-@item VideoTag @var{tag} (@emph{encoding,video})
-Set video tag.
-
-@item VideoHighQuality (@emph{encoding,video})
-@item Video4MotionVector (@emph{encoding,video})
-
-@item BitExact (@emph{encoding,video})
-Set bitexact encoding flag.
-
-@item IdctSimple (@emph{encoding,video})
-Set simple IDCT algorithm.
-
-@item Qscale @var{n} (@emph{encoding,video})
-Enable constant quality encoding, and set video qscale (quantization
-scale) value, expressed in @var{n} QP units.
-
-@item VideoQMin @var{n} (@emph{encoding,video})
-@item VideoQMax @var{n} (@emph{encoding,video})
-Set video qmin/qmax.
-
-@item VideoQDiff @var{integer} (@emph{encoding,video})
-Set video @option{qdiff} encoding option.
-
-@item LumiMask @var{float} (@emph{encoding,video})
-@item DarkMask @var{float} (@emph{encoding,video})
-Set @option{lumi_mask}/@option{dark_mask} encoding options.
-
-@item AVOptionVideo [@var{codec}:]@var{option} @var{value} (@emph{encoding,video})
-Set generic or private option for video stream.
-Private option must be prefixed with codec name or codec must be defined before.
-
-@item AVPresetVideo @var{preset} (@emph{encoding,video})
-Set preset for video stream.
-
-@var{preset} must be the path of a preset file.
-@end table
-
-@subsection Server status stream
-
-A server status stream is a special stream which is used to show
-statistics about the @command{ffserver} operations.
-
-It must be specified setting the option @option{Format} to
-@samp{status}.
-
-@section Redirect section
-
-A redirect section specifies where to redirect the requested URL to
-another page.
-
-A redirect section must be introduced by the line:
-@example
-<Redirect NAME>
-@end example
-
-where @var{NAME} is the name of the page which should be redirected.
-
-It only accepts the option @option{URL}, which specify the redirection
-URL.
-
-@chapter Stream examples
-
-@itemize
-@item
-Multipart JPEG
-@example
-<Stream test.mjpg>
-Feed feed1.ffm
-Format mpjpeg
-VideoFrameRate 2
-VideoIntraOnly
-NoAudio
-Strict -1
-</Stream>
-@end example
-
-@item
-Single JPEG
-@example
-<Stream test.jpg>
-Feed feed1.ffm
-Format jpeg
-VideoFrameRate 2
-VideoIntraOnly
-VideoSize 352x240
-NoAudio
-Strict -1
-</Stream>
-@end example
-
-@item
-Flash
-@example
-<Stream test.swf>
-Feed feed1.ffm
-Format swf
-VideoFrameRate 2
-VideoIntraOnly
-NoAudio
-</Stream>
-@end example
-
-@item
-ASF compatible
-@example
-<Stream test.asf>
-Feed feed1.ffm
-Format asf
-VideoFrameRate 15
-VideoSize 352x240
-VideoBitRate 256
-VideoBufferSize 40
-VideoGopSize 30
-AudioBitRate 64
-StartSendOnKey
-</Stream>
-@end example
-
-@item
-MP3 audio
-@example
-<Stream test.mp3>
-Feed feed1.ffm
-Format mp2
-AudioCodec mp3
-AudioBitRate 64
-AudioChannels 1
-AudioSampleRate 44100
-NoVideo
-</Stream>
-@end example
-
-@item
-Ogg Vorbis audio
-@example
-<Stream test.ogg>
-Feed feed1.ffm
-Metadata title "Stream title"
-AudioBitRate 64
-AudioChannels 2
-AudioSampleRate 44100
-NoVideo
-</Stream>
-@end example
-
-@item
-Real with audio only at 32 kbits
-@example
-<Stream test.ra>
-Feed feed1.ffm
-Format rm
-AudioBitRate 32
-NoVideo
-</Stream>
-@end example
-
-@item
-Real with audio and video at 64 kbits
-@example
-<Stream test.rm>
-Feed feed1.ffm
-Format rm
-AudioBitRate 32
-VideoBitRate 128
-VideoFrameRate 25
-VideoGopSize 25
-</Stream>
-@end example
-
-@item
-For stream coming from a file: you only need to set the input filename
-and optionally a new format.
-
-@example
-<Stream file.rm>
-File "/usr/local/httpd/htdocs/tlive.rm"
-NoAudio
-</Stream>
-@end example
-
-@example
-<Stream file.asf>
-File "/usr/local/httpd/htdocs/test.asf"
-NoAudio
-Metadata author "Me"
-Metadata copyright "Super MegaCorp"
-Metadata title "Test stream from disk"
-Metadata comment "Test comment"
-</Stream>
-@end example
-@end itemize
-
 @c man end
 
 @include config.texi

doc/fftools-common-opts.texi

@@ -3,7 +3,7 @@ representing a number as input, which may be followed by one of the SI
 unit prefixes, for example: 'K', 'M', or 'G'.
 
 If 'i' is appended to the SI unit prefix, the complete prefix will be
-interpreted as a unit prefix for binary multiples, which are based on
+interpreted as a unit prefix for binary multiplies, which are based on
 powers of 1024 instead of powers of 1000. Appending 'B' to the SI unit
 prefix multiplies the value by 8. This allows using, for example:
 'KB', 'MiB', 'G' and 'B' as number suffixes.
@@ -44,15 +44,8 @@ streams of this type.
 If @var{stream_index} is given, then it matches the stream with number @var{stream_index}
 in the program with the id @var{program_id}. Otherwise, it matches all streams in the
 program.
-@item #@var{stream_id} or i:@var{stream_id}
-Match the stream by stream id (e.g. PID in MPEG-TS container).
-@item m:@var{key}[:@var{value}]
-Matches streams with the metadata tag @var{key} having the specified value. If
-@var{value} is not given, matches streams that contain the given tag with any
-value.
-
-Note that in @command{ffmpeg}, matching by metadata will only work properly for
-input files.
+@item #@var{stream_id}
+Matches the stream by a format-specific ID.
 @end table
 
 @section Generic options
@@ -103,10 +96,7 @@ Print detailed information about the filter name @var{filter_name}. Use the
 Show version.
 
 @item -formats
-Show available formats (including devices).
-
-@item -devices
-Show available devices.
+Show available formats.
 
 @item -codecs
 Show all codecs known to libavcodec.
@@ -141,22 +131,6 @@ Show channel names and standard channel layouts.
 @item -colors
 Show recognized color names.
 
-@item -sources @var{device}[,@var{opt1}=@var{val1}[,@var{opt2}=@var{val2}]...]
-Show autodetected sources of the intput device.
-Some devices may provide system-dependent source names that cannot be autodetected.
-The returned list cannot be assumed to be always complete.
-@example
-ffmpeg -sources pulse,server=192.168.0.4
-@end example
-
-@item -sinks @var{device}[,@var{opt1}=@var{val1}[,@var{opt2}=@var{val2}]...]
-Show autodetected sinks of the output device.
-Some devices may provide system-dependent sink names that cannot be autodetected.
-The returned list cannot be assumed to be always complete.
-@example
-ffmpeg -sinks pulse,server=192.168.0.4
-@end example
-
 @item -loglevel [repeat+]@var{loglevel} | -v [repeat+]@var{loglevel}
 Set the logging level used by the library.
 Adding "repeat+" indicates that repeated log output should not be compressed
@@ -165,27 +139,27 @@ omitted. "repeat" can also be used alone.
 If "repeat" is used alone, and with no prior loglevel set, the default
 loglevel will be used. If multiple loglevel parameters are given, using
 'repeat' will not change the loglevel.
-@var{loglevel} is a string or a number containing one of the following values:
+@var{loglevel} is a number or a string containing one of the following values:
 @table @samp
-@item quiet, -8
+@item quiet
 Show nothing at all; be silent.
-@item panic, 0
+@item panic
 Only show fatal errors which could lead the process to crash, such as
 and assert failure. This is not currently used for anything.
-@item fatal, 8
+@item fatal
 Only show fatal errors. These are errors after which the process absolutely
 cannot continue after.
-@item error, 16
+@item error
 Show all errors, including ones which can be recovered from.
-@item warning, 24
+@item warning
 Show all warnings and errors. Any message related to possibly
 incorrect or unexpected events will be shown.
-@item info, 32
+@item info
 Show informative messages during processing. This is in addition to
 warnings and errors. This is the default value.
-@item verbose, 40
+@item verbose
 Same as @code{info}, except more verbose.
-@item debug, 48
+@item debug
 Show everything, including debugging information.
 @end table
 
@@ -204,39 +178,22 @@ directory.
 This file can be useful for bug reports.
 It also implies @code{-loglevel verbose}.
 
-Setting the environment variable @env{FFREPORT} to any value has the
+Setting the environment variable @code{FFREPORT} to any value has the
 same effect. If the value is a ':'-separated key=value sequence, these
-options will affect the report; option values must be escaped if they
+options will affect the report; options values must be escaped if they
 contain special characters or the options delimiter ':' (see the
-``Quoting and escaping'' section in the ffmpeg-utils manual).
-
-The following options are recognized:
+``Quoting and escaping'' section in the ffmpeg-utils manual). The
+following option is recognized:
 @table @option
 @item file
 set the file name to use for the report; @code{%p} is expanded to the name
 of the program, @code{%t} is expanded to a timestamp, @code{%%} is expanded
 to a plain @code{%}
-@item level
-set the log verbosity level using a numerical value (see @code{-loglevel}).
 @end table
 
-For example, to output a report to a file named @file{ffreport.log}
-using a log level of @code{32} (alias for log level @code{info}):
-
-@example
-FFREPORT=file=ffreport.log:level=32 ffmpeg -i input output
-@end example
-
 Errors in parsing the environment variable are not fatal, and will not
 appear in the report.
 
-@item -hide_banner
-Suppress printing banner.
-
-All FFmpeg tools will normally show a copyright notice, build options
-and library versions. This option can be used to suppress printing
-this information.
-
 @item -cpuflags flags (@emph{global})
 Allows setting and clearing cpu flags. This option is intended
 for testing. Do not use it unless you know what you're doing.
@@ -261,14 +218,10 @@ Possible flags for this option are:
 @item sse4.1
 @item sse4.2
 @item avx
-@item avx2
 @item xop
-@item fma3
 @item fma4
 @item 3dnow
 @item 3dnowext
-@item bmi1
-@item bmi2
 @item cmov
 @end table
 @item ARM
@@ -279,13 +232,6 @@ Possible flags for this option are:
 @item vfp
 @item vfpv3
 @item neon
-@item setend
-@end table
-@item AArch64
-@table @samp
-@item armv8
-@item vfp
-@item neon
 @end table
 @item PowerPC
 @table @samp
@@ -304,43 +250,6 @@ Possible flags for this option are:
 @end table
 @end table
 
-@item -opencl_bench
-This option is used to benchmark all available OpenCL devices and print the
-results. This option is only available when FFmpeg has been compiled with
-@code{--enable-opencl}.
-
-When FFmpeg is configured with @code{--enable-opencl}, the options for the
-global OpenCL context are set via @option{-opencl_options}. See the
-"OpenCL Options" section in the ffmpeg-utils manual for the complete list of
-supported options. Amongst others, these options include the ability to select
-a specific platform and device to run the OpenCL code on. By default, FFmpeg
-will run on the first device of the first platform. While the options for the
-global OpenCL context provide flexibility to the user in selecting the OpenCL
-device of their choice, most users would probably want to select the fastest
-OpenCL device for their system.
-
-This option assists the selection of the most efficient configuration by
-identifying the appropriate device for the user's system. The built-in
-benchmark is run on all the OpenCL devices and the performance is measured for
-each device. The devices in the results list are sorted based on their
-performance with the fastest device listed first. The user can subsequently
-invoke @command{ffmpeg} using the device deemed most appropriate via
-@option{-opencl_options} to obtain the best performance for the OpenCL
-accelerated code.
-
-Typical usage to use the fastest OpenCL device involve the following steps.
-
-Run the command:
-@example
-ffmpeg -opencl_bench
-@end example
-Note down the platform ID (@var{pidx}) and device ID (@var{didx}) of the first
-i.e. fastest device in the list.
-Select the platform and device using the command:
-@example
-ffmpeg -opencl_options platform_idx=@var{pidx}:device_idx=@var{didx} ...
-@end example
-
 @item -opencl_options options (@emph{global})
 Set OpenCL environment options. This option is only available when
 FFmpeg has been compiled with @code{--enable-opencl}.

doc/formats.texi

@@ -23,7 +23,7 @@ Reduce buffering.
 
 @item probesize @var{integer} (@emph{input})
 Set probing size in bytes, i.e. the size of the data to analyze to get
-stream information. A higher value will enable detecting more
+stream information. A higher value will allow to detect more
 information in case it is dispersed into the stream, but will increase
 latency. Must be an integer not lesser than 32. It is 5000000 by default.
 
@@ -55,10 +55,6 @@ Do not merge side data.
 Enable RTP MP4A-LATM payload.
 @item nobuffer
 Reduce the latency introduced by optional buffering
-@item bitexact
-Only write platform-, build- and time-independent data.
-This ensures that file and data checksums are reproducible and match between
-platforms. Its primary use is for regression testing.
 @end table
 
 @item seek2any @var{integer} (@emph{input})
@@ -67,7 +63,7 @@ Default is 0.
 
 @item analyzeduration @var{integer} (@emph{input})
 Specify how many microseconds are analyzed to probe the input. A
-higher value will enable detecting more accurate information, but will
+higher value will allow to detect more accurate information, but will
 increase latency. It defaults to 5,000,000 microseconds = 5 seconds.
 
 @item cryptokey @var{hexadecimal string} (@emph{input})
@@ -129,26 +125,18 @@ Consider things that a sane encoder should not do as an error.
 Use wallclock as timestamps.
 
 @item avoid_negative_ts @var{integer} (@emph{output})
-
-Possible values:
-@table @samp
-@item make_non_negative
-Shift timestamps to make them non-negative.
-Also note that this affects only leading negative timestamps, and not
-non-monotonic negative timestamps.
-@item make_zero
-Shift timestamps so that the first timestamp is 0.
-@item auto (default)
-Enables shifting when required by the target format.
-@item disabled
-Disables shifting of timestamp.
-@end table
+Shift timestamps to make them non-negative. A value of 1 enables shifting,
+a value of 0 disables it, the default value of -1 enables shifting
+when required by the target format.
 
 When shifting is enabled, all output timestamps are shifted by the
 same amount. Audio, video, and subtitles desynching and relative
 timestamp differences are preserved compared to how they would have
 been without shifting.
 
+Also note that this affects only leading negative timestamps, and not
+non-monotonic negative timestamps.
+
 @item skip_initial_bytes @var{integer} (@emph{input})
 Set number of bytes to skip before reading header and frames if set to 1.
 Default is 0.
@@ -160,30 +148,6 @@ Correct single timestamp overflows if set to 1. Default is 1.
 Flush the underlying I/O stream after each packet. Default 1 enables it, and
 has the effect of reducing the latency; 0 disables it and may slightly
 increase performance in some cases.
-
-@item output_ts_offset @var{offset} (@emph{output})
-Set the output time offset.
-
-@var{offset} must be a time duration specification,
-see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
-
-The offset is added by the muxer to the output timestamps.
-
-Specifying a positive offset means that the corresponding streams are
-delayed bt the time duration specified in @var{offset}. Default value
-is @code{0} (meaning that no offset is applied).
-
-@item format_whitelist @var{list} (@emph{input})
-"," separated List of allowed demuxers. By default all are allowed.
-
-@item dump_separator @var{string} (@emph{input})
-Separator used to separate the fields printed on the command line about the
-Stream parameters.
-For example to separate the fields with newlines and indention:
-@example
-ffprobe -dump_separator "
-                          "  -i ~/videos/matrixbench_mpeg2.mpg
-@end example
 @end table
 
 @c man end FORMAT OPTIONS
@@ -219,10 +183,6 @@ The exact semantics of stream specifiers is defined by the
 @code{avformat_match_stream_specifier()} function declared in the
 @file{libavformat/avformat.h} header.
 
-@ifclear config-writeonly
 @include demuxers.texi
-@end ifclear
-@ifclear config-readonly
 @include muxers.texi
-@end ifclear
 @include metadata.texi

doc/general.texi

@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8
 
 @settitle General Documentation
 @titlepage
@@ -109,14 +108,6 @@ Go to @url{http://www.wavpack.com/} and follow the instructions for
 installing the library. Then pass @code{--enable-libwavpack} to configure to
 enable it.
 
-@section OpenH264
-
-FFmpeg can make use of the OpenH264 library for H.264 encoding.
-
-Go to @url{http://www.openh264.org/} and follow the instructions for
-installing the library. Then pass @code{--enable-libopenh264} to configure to
-enable it.
-
 @section x264
 
 FFmpeg can make use of the x264 library for H.264 encoding.
@@ -131,20 +122,6 @@ x264 is under the GNU Public License Version 2 or later
 details), you must upgrade FFmpeg's license to GPL in order to use it.
 @end float
 
-@section x265
-
-FFmpeg can make use of the x265 library for HEVC encoding.
-
-Go to @url{http://x265.org/developers.html} and follow the instructions
-for installing the library. Then pass @code{--enable-libx265} to configure
-to enable it.
-
-@float NOTE
-x265 is under the GNU Public License Version 2 or later
-(see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for
-details), you must upgrade FFmpeg's license to GPL in order to use it.
-@end float
-
 @section libilbc
 
 iLBC is a narrowband speech codec that has been made freely available
@@ -152,7 +129,7 @@ by Google as part of the WebRTC project. libilbc is a packaging friendly
 copy of the iLBC codec. FFmpeg can make use of the libilbc library for
 iLBC encoding and decoding.
 
-Go to @url{https://github.com/TimothyGu/libilbc} and follow the instructions for
+Go to @url{https://github.com/dekkers/libilbc} and follow the instructions for
 installing the library. Then pass @code{--enable-libilbc} to configure to
 enable it.
 
@@ -171,27 +148,6 @@ libzvbi is licensed under the GNU General Public License Version 2 or later
 you must upgrade FFmpeg's license to GPL in order to use it.
 @end float
 
-@section AviSynth
-
-FFmpeg can read AviSynth scripts as input. To enable support, pass
-@code{--enable-avisynth} to configure.  The correct headers are
-included in compat/avisynth/, which allows the user to enable support
-without needing to search for these headers themselves.
-
-For Windows, supported AviSynth variants are
-@url{http://avisynth.nl, AviSynth 2.5 or 2.6} for 32-bit builds and
-@url{http://avs-plus.net, AviSynth+ 0.1} for 32-bit and 64-bit builds.
-
-For Linux and OS X, the supported AviSynth variant is
-@url{https://github.com/avxsynth/avxsynth, AvxSynth}.
-
-@float NOTE
-AviSynth and AvxSynth are loaded dynamically.  Distributors can build FFmpeg
-with @code{--enable-avisynth}, and the binaries will work regardless of the
-end user having AviSynth or AvxSynth installed - they'll only need to be
-installed to use AviSynth scripts (obviously).
-@end float
-
 
 @chapter Supported File Formats, Codecs or Features
 
@@ -252,11 +208,8 @@ library:
     @tab Used in the game Cyberia from Interplay.
 @item Delphine Software International CIN @tab   @tab X
     @tab Multimedia format used by Delphine Software games.
-@item Digital Speech Standard (DSS) @tab   @tab X
-@item Canopus HQX               @tab   @tab X
 @item CD+G                      @tab   @tab X
     @tab Video format used by CD+G karaoke disks
-@item Phantom Cine              @tab   @tab X
 @item Commodore CDXL            @tab   @tab X
     @tab Amiga CD video format
 @item Core Audio Format         @tab X @tab X
@@ -270,7 +223,6 @@ library:
 @item Deluxe Paint Animation    @tab   @tab X
 @item DFA                       @tab   @tab X
     @tab This format is used in Chronomaster game
-@item DSD Stream File (DSF)     @tab   @tab X
 @item DV video                  @tab X @tab X
 @item DXA                       @tab   @tab X
     @tab This format is used in the non-Windows version of the Feeble Files
@@ -297,8 +249,6 @@ library:
 @item GXF                       @tab X @tab X
     @tab General eXchange Format SMPTE 360M, used by Thomson Grass Valley
          playout servers.
-@item HNM @tab   @tab X
-    @tab Only version 4 supported, used in some games from Cryo Interactive
 @item iCEDraw File              @tab   @tab X
 @item ICO                       @tab X @tab X
     @tab Microsoft Windows ICO
@@ -321,11 +271,9 @@ library:
     @tab Used by Linux Media Labs MPEG-4 PCI boards
 @item LOAS                      @tab   @tab X
     @tab contains LATM multiplexed AAC audio
-@item LRC                       @tab X @tab X
 @item LVF                       @tab   @tab X
 @item LXF                       @tab   @tab X
     @tab VR native stream format, used by Leitch/Harris' video servers.
-@item Magic Lantern Video (MLV) @tab   @tab X
 @item Matroska                  @tab X @tab X
 @item Matroska audio            @tab X @tab
 @item FFmpeg metadata           @tab X @tab X
@@ -351,8 +299,6 @@ library:
     @tab also known as DVB Transport Stream
 @item MPEG-4                    @tab X @tab X
     @tab MPEG-4 is a variant of QuickTime.
-@item Mirillis FIC video        @tab   @tab X
-    @tab No cursor rendering.
 @item MIME multipart JPEG       @tab X @tab
 @item MSN TCP webcam            @tab   @tab X
     @tab Used by MSN Messenger webcam streams.
@@ -392,7 +338,6 @@ library:
 @item raw H.261                 @tab X @tab X
 @item raw H.263                 @tab X @tab X
 @item raw H.264                 @tab X @tab X
-@item raw HEVC                  @tab X @tab X
 @item raw Ingenient MJPEG       @tab   @tab X
 @item raw MJPEG                 @tab X @tab X
 @item raw MLP                   @tab   @tab X
@@ -465,7 +410,6 @@ library:
 @item Sony Wave64 (W64)         @tab X @tab X
 @item SoX native format         @tab X @tab X
 @item SUN AU format             @tab X @tab X
-@item SUP raw PGS subtitles     @tab   @tab X
 @item Text files                @tab   @tab X
 @item THP                       @tab   @tab X
     @tab Used on the Nintendo GameCube.
@@ -504,13 +448,11 @@ following image formats are supported:
 @item Name @tab Encoding @tab Decoding @tab Comments
 @item .Y.U.V       @tab X @tab X
     @tab one raw file per component
-@item Alias PIX    @tab X @tab X
-    @tab Alias/Wavefront PIX image format
 @item animated GIF @tab X @tab X
 @item BMP          @tab X @tab X
     @tab Microsoft BMP image
-@item BRender PIX  @tab   @tab X
-    @tab Argonaut BRender 3D engine image format.
+@item PIX          @tab   @tab X
+    @tab PIX is an image format used in the Argonaut BRender engine.
 @item DPX          @tab X @tab X
     @tab Digital Picture Exchange
 @item EXR          @tab   @tab X
@@ -546,8 +488,8 @@ following image formats are supported:
     @tab YUV, JPEG and some extension is not supported yet.
 @item Truevision Targa  @tab X @tab X
     @tab Targa (.TGA) image format
-@item WebP         @tab E @tab X
-    @tab WebP image format, encoding supported through external library libwebp
+@item WebP         @tab @tab X
+    @tab WebP image format
 @item XBM  @tab X @tab X
     @tab X BitMap image format
 @item XFace @tab X @tab X
@@ -664,10 +606,7 @@ following image formats are supported:
 @item H.263 / H.263-1996     @tab  X  @tab  X
 @item H.263+ / H.263-1998 / H.263 version 2  @tab  X  @tab  X
 @item H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10  @tab  E  @tab  X
-    @tab encoding supported through external library libx264 and OpenH264
-@item HEVC                   @tab  X  @tab  X
-    @tab encoding supported through the external library libx265
-@item HNM version 4          @tab     @tab  X
+    @tab encoding supported through external library libx264
 @item HuffYUV                @tab  X  @tab  X
 @item HuffYUV FFmpeg variant @tab  X  @tab  X
 @item IBM Ultimotion         @tab     @tab  X
@@ -698,8 +637,8 @@ following image formats are supported:
 @item LCL (LossLess Codec Library) MSZH  @tab     @tab  X
 @item LCL (LossLess Codec Library) ZLIB  @tab  E  @tab  E
 @item LOCO                   @tab     @tab  X
-@item LucasArts SANM/Smush   @tab     @tab  X
-    @tab Used in LucasArts games / SMUSH animations.
+@item LucasArts Smush        @tab     @tab  X
+    @tab Used in LucasArts games.
 @item lossless MJPEG         @tab  X  @tab  X
 @item Microsoft ATC Screen   @tab     @tab  X
     @tab Also known as Microsoft Screen 3.
@@ -719,6 +658,7 @@ following image formats are supported:
 @item Mobotix MxPEG video    @tab     @tab  X
 @item Motion Pixels video    @tab     @tab  X
 @item MPEG-1 video           @tab  X  @tab  X
+@item MPEG-1/2 video XvMC (X-Video Motion Compensation)  @tab     @tab  X
 @item MPEG-2 video           @tab  X  @tab  X
 @item MPEG-4 part 2          @tab  X  @tab  X
     @tab libxvidcore can be used alternatively for encoding.
@@ -734,8 +674,6 @@ following image formats are supported:
     @tab fourcc: VP50
 @item On2 VP6                @tab     @tab  X
     @tab fourcc: VP60,VP61,VP62
-@item On2 VP7                @tab     @tab  X
-    @tab fourcc: VP70,VP71
 @item VP8                    @tab  E  @tab  X
     @tab fourcc: VP80, encoding supported through external library libvpx
 @item VP9                    @tab  E  @tab  X
@@ -765,11 +703,11 @@ following image formats are supported:
     @tab Texture dictionaries used by the Renderware Engine.
 @item RL2 video              @tab     @tab  X
     @tab used in some games by Entertainment Software Partners
+@item SGI RLE 8-bit          @tab     @tab  X
 @item Sierra VMD video       @tab     @tab  X
     @tab Used in Sierra VMD files.
 @item Silicon Graphics Motion Video Compressor 1 (MVC1)  @tab     @tab  X
 @item Silicon Graphics Motion Video Compressor 2 (MVC2)  @tab     @tab  X
-@item Silicon Graphics RLE 8-bit video  @tab     @tab  X
 @item Smacker video          @tab     @tab  X
     @tab Video encoding used in Smacker.
 @item SMPTE VC-1             @tab     @tab  X
@@ -835,7 +773,7 @@ following image formats are supported:
     @tab encoding supported through external library libaacplus
 @item AAC                    @tab  E  @tab  X
     @tab encoding supported through external library libfaac and libvo-aacenc
-@item AC-3                   @tab IX  @tab  IX
+@item AC-3                   @tab IX  @tab  X
 @item ADPCM 4X Movie         @tab     @tab  X
 @item ADPCM CDROM XA         @tab     @tab  X
 @item ADPCM Creative Technology @tab     @tab  X
@@ -879,8 +817,6 @@ following image formats are supported:
 @item ADPCM Sound Blaster Pro 2-bit  @tab     @tab  X
 @item ADPCM Sound Blaster Pro 2.6-bit  @tab     @tab  X
 @item ADPCM Sound Blaster Pro 4-bit  @tab     @tab  X
-@item ADPCM VIMA
-    @tab Used in LucasArts SMUSH animations.
 @item ADPCM Westwood Studios IMA @tab     @tab  X
     @tab Used in Westwood Studios games like Command and Conquer.
 @item ADPCM Yamaha           @tab  X  @tab  X
@@ -893,14 +829,12 @@ following image formats are supported:
     @tab QuickTime fourcc 'alac'
 @item ATRAC1                 @tab     @tab  X
 @item ATRAC3                 @tab     @tab  X
-@item ATRAC3+                @tab     @tab  X
 @item Bink Audio             @tab     @tab  X
     @tab Used in Bink and Smacker files in many games.
 @item CELT                   @tab     @tab  E
     @tab decoding supported through external library libcelt
 @item Delphine Software International CIN audio  @tab     @tab  X
     @tab Codec used in Delphine Software International games.
-@item Digital Speech Standard - Standard Play mode (DSS SP) @tab     @tab  X
 @item Discworld II BMV Audio @tab     @tab  X
 @item COOK                   @tab     @tab  X
     @tab All versions except 5.1 are supported.
@@ -914,10 +848,6 @@ following image formats are supported:
 @item DPCM Sol               @tab     @tab  X
 @item DPCM Xan               @tab     @tab  X
     @tab Used in Origin's Wing Commander IV AVI files.
-@item DSD (Direct Stream Digitial), least significant bit first  @tab  @tab  X
-@item DSD (Direct Stream Digitial), most significant bit first   @tab  @tab  X
-@item DSD (Direct Stream Digitial), least significant bit first, planar  @tab  @tab  X
-@item DSD (Direct Stream Digitial), most significant bit first, planar   @tab  @tab  X
 @item DSP Group TrueSpeech   @tab     @tab  X
 @item DV audio               @tab     @tab  X
 @item Enhanced AC-3          @tab  X  @tab  X
@@ -940,16 +870,15 @@ following image formats are supported:
 @item Monkey's Audio         @tab     @tab  X
 @item MP1 (MPEG audio layer 1)  @tab     @tab IX
 @item MP2 (MPEG audio layer 2)  @tab IX  @tab IX
-    @tab encoding supported also through external library TwoLAME
+    @tab libtwolame can be used alternatively for encoding.
 @item MP3 (MPEG audio layer 3)  @tab  E  @tab IX
     @tab encoding supported through external library LAME, ADU MP3 and MP3onMP4 also supported
 @item MPEG-4 Audio Lossless Coding (ALS)  @tab     @tab  X
 @item Musepack SV7           @tab     @tab  X
 @item Musepack SV8           @tab     @tab  X
 @item Nellymoser Asao        @tab  X  @tab  X
-@item On2 AVC (Audio for Video Codec) @tab     @tab  X
-@item Opus                   @tab  E  @tab  X
-    @tab encoding supported through external library libopus
+@item Opus                   @tab  E  @tab  E
+    @tab supported through external library libopus
 @item PCM A-law              @tab  X  @tab  X
 @item PCM mu-law             @tab  X  @tab  X
 @item PCM signed 8-bit planar  @tab  X  @tab  X
@@ -1010,6 +939,7 @@ following image formats are supported:
 @item Vorbis                 @tab  E  @tab  X
     @tab A native but very primitive encoder exists.
 @item Voxware MetaSound      @tab     @tab  X
+    @tab imperfect and incomplete support
 @item WavPack                @tab  X  @tab  X
 @item Westwood Audio (SND1)  @tab     @tab  X
 @item Windows Media Audio 1  @tab  X  @tab  X
@@ -1043,7 +973,6 @@ performance on systems without hardware floating point support).
 @item PJS (Phoenix)    @tab   @tab X @tab   @tab X
 @item RealText         @tab   @tab X @tab   @tab X
 @item SAMI             @tab   @tab X @tab   @tab X
-@item Spruce format (STL) @tab   @tab X @tab   @tab X
 @item SSA/ASS          @tab X @tab X @tab X @tab X
 @item SubRip (SRT)     @tab X @tab X @tab X @tab X
 @item SubViewer v1     @tab   @tab X @tab   @tab X
@@ -1051,7 +980,7 @@ performance on systems without hardware floating point support).
 @item TED Talks captions @tab @tab X @tab   @tab X
 @item VobSub (IDX+SUB) @tab   @tab X @tab   @tab X
 @item VPlayer          @tab   @tab X @tab   @tab X
-@item WebVTT           @tab X @tab X @tab X @tab X
+@item WebVTT           @tab X @tab X @tab   @tab X
 @item XSUB             @tab   @tab   @tab X @tab X
 @end multitable
 
@@ -1064,12 +993,10 @@ performance on systems without hardware floating point support).
 @multitable @columnfractions .4 .1
 @item Name         @tab Support
 @item file         @tab X
-@item FTP          @tab X
 @item Gopher       @tab X
 @item HLS          @tab X
 @item HTTP         @tab X
 @item HTTPS        @tab X
-@item Icecast      @tab X
 @item MMSH         @tab X
 @item MMST         @tab X
 @item pipe         @tab X
@@ -1080,9 +1007,7 @@ performance on systems without hardware floating point support).
 @item RTMPTE       @tab X
 @item RTMPTS       @tab X
 @item RTP          @tab X
-@item SAMBA        @tab E
 @item SCTP         @tab X
-@item SFTP         @tab E
 @item TCP          @tab X
 @item TLS          @tab X
 @item UDP          @tab X
@@ -1102,19 +1027,17 @@ performance on systems without hardware floating point support).
 @item caca              @tab        @tab X
 @item DV1394            @tab X      @tab
 @item Lavfi virtual device @tab X   @tab
-@item Linux framebuffer @tab X      @tab X
+@item Linux framebuffer @tab X      @tab
 @item JACK              @tab X      @tab
 @item LIBCDIO           @tab X
 @item LIBDC1394         @tab X      @tab
 @item OpenAL            @tab X
-@item OpenGL            @tab        @tab X
 @item OSS               @tab X      @tab X
-@item PulseAudio        @tab X      @tab X
+@item Pulseaudio        @tab X      @tab
 @item SDL               @tab        @tab X
 @item Video4Linux2      @tab X      @tab X
 @item VfW capture       @tab X      @tab
 @item X11 grabbing      @tab X      @tab
-@item Win32 grabbing    @tab X      @tab
 @end multitable
 
 @code{X} means that input/output is supported.

doc/git-howto.texi

@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8
 
 @settitle Using git to develop FFmpeg
 
@@ -300,7 +299,7 @@ the current branch history.
 git commit --amend
 @end example
 
-allows one to amend the last commit details quickly.
+allows to amend the last commit details quickly.
 
 @example
 git rebase -i origin/master

doc/git-howto.txt

@@ -0,0 +1,273 @@
+
+About Git write access:
+~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+
+Before everything else, you should know how to use GIT properly.
+Luckily Git comes with excellent documentation.
+
+  git --help
+  man git
+
+shows you the available subcommands,
+
+  git <command> --help
+  man git-<command>
+
+shows information about the subcommand <command>.
+
+The most comprehensive manual is the website Git Reference
+
+http://gitref.org/
+
+For more information about the Git project, visit
+
+http://git-scm.com/
+
+Consult these resources whenever you have problems, they are quite exhaustive.
+
+You do not need a special username or password.
+All you need is to provide a ssh public key to the Git server admin.
+
+What follows now is a basic introduction to Git and some FFmpeg-specific
+guidelines. Read it at least once, if you are granted commit privileges to the
+FFmpeg project you are expected to be familiar with these rules.
+
+
+
+I. BASICS:
+==========
+
+0. Get GIT:
+
+  Most distributions have a git package, if not
+  You can get git from http://git-scm.com/
+
+
+1. Cloning the source tree:
+
+    git clone git://source.ffmpeg.org/ffmpeg <target>
+
+  This will put the FFmpeg sources into the directory <target>.
+
+    git clone git@source.ffmpeg.org:ffmpeg <target>
+
+  This will put the FFmpeg sources into the directory <target> and let
+  you push back your changes to the remote repository.
+
+
+2. Updating the source tree to the latest revision:
+
+    git pull (--ff-only)
+
+  pulls in the latest changes from the tracked branch. The tracked branch
+  can be remote. By default the master branch tracks the branch master in
+  the remote origin.
+  Caveat: Since merge commits are forbidden at least for the initial
+          months of git --ff-only or --rebase (see below) are recommended.
+          --ff-only will fail and not create merge commits if your branch
+          has diverged (has a different history) from the tracked branch.
+
+2.a Rebasing your local branches:
+
+    git pull --rebase
+
+  fetches the changes from the main repository and replays your local commits
+  over it. This is required to keep all your local changes at the top of
+  FFmpeg's master tree. The master tree will reject pushes with merge commits.
+
+
+3. Adding/removing files/directories:
+
+    git add [-A] <filename/dirname>
+    git rm [-r] <filename/dirname>
+
+  GIT needs to get notified of all changes you make to your working
+  directory that makes files appear or disappear.
+  Line moves across files are automatically tracked.
+
+
+4. Showing modifications:
+
+    git diff <filename(s)>
+
+  will show all local modifications in your working directory as unified diff.
+
+
+5. Inspecting the changelog:
+
+    git log <filename(s)>
+
+  You may also use the graphical tools like gitview or gitk or the web
+  interface available at http://source.ffmpeg.org
+
+6. Checking source tree status:
+
+    git status
+
+  detects all the changes you made and lists what actions will be taken in case
+  of a commit (additions, modifications, deletions, etc.).
+
+
+7. Committing:
+
+    git diff --check
+
+  to double check your changes before committing them to avoid trouble later
+  on. All experienced developers do this on each and every commit, no matter
+  how small.
+  Every one of them has been saved from looking like a fool by this many times.
+  It's very easy for stray debug output or cosmetic modifications to slip in,
+  please avoid problems through this extra level of scrutiny.
+
+  For cosmetics-only commits you should get (almost) empty output from
+
+    git diff -w -b <filename(s)>
+
+  Also check the output of
+
+    git status
+
+  to make sure you don't have untracked files or deletions.
+
+    git add [-i|-p|-A] <filenames/dirnames>
+
+  Make sure you have told git your name and email address, e.g. by running
+    git config --global user.name "My Name"
+    git config --global user.email my@email.invalid
+  (--global to set the global configuration for all your git checkouts).
+
+  Git will select the changes to the files for commit. Optionally you can use
+  the interactive or the patch mode to select hunk by hunk what should be
+  added to the commit.
+
+    git commit
+
+  Git will commit the selected changes to your current local branch.
+
+  You will be prompted for a log message in an editor, which is either
+  set in your personal configuration file through
+
+    git config core.editor
+
+  or set by one of the following environment variables:
+  GIT_EDITOR, VISUAL or EDITOR.
+
+  Log messages should be concise but descriptive. Explain why you made a change,
+  what you did will be obvious from the changes themselves most of the time.
+  Saying just "bug fix" or "10l" is bad. Remember that people of varying skill
+  levels look at and educate themselves while reading through your code. Don't
+  include filenames in log messages, Git provides that information.
+
+  Possibly make the commit message have a terse, descriptive first line, an
+  empty line and then a full description. The first line will be used to name
+  the patch by git format-patch.
+
+
+8. Renaming/moving/copying files or contents of files:
+
+  Git automatically tracks such changes, making those normal commits.
+
+    mv/cp path/file otherpath/otherfile
+
+    git add [-A] .
+
+    git commit
+
+  Do not move, rename or copy files of which you are not the maintainer without
+  discussing it on the mailing list first!
+
+9. Reverting broken commits
+
+    git revert <commit>
+
+  git revert will generate a revert commit. This will not make the faulty
+  commit disappear from the history.
+
+    git reset <commit>
+
+  git reset will uncommit the changes till <commit> rewriting the current
+  branch history.
+
+    git commit --amend
+
+  allows to amend the last commit details quickly.
+
+    git rebase -i origin/master
+
+  will replay local commits over the main repository allowing to edit,
+  merge or remove some of them in the process.
+
+  Note that the reset, commit --amend and rebase rewrite history, so you
+  should use them ONLY on your local or topic branches.
+
+  The main repository will reject those changes.
+
+10. Preparing a patchset.
+
+    git format-patch <commit> [-o directory]
+
+  will generate a set of patches for each commit between <commit> and
+  current HEAD. E.g.
+
+    git format-patch origin/master
+
+  will generate patches for all commits on current branch which are not
+  present in upstream.
+  A useful shortcut is also
+
+    git format-patch -n
+
+  which will generate patches from last n commits.
+  By default the patches are created in the current directory.
+
+11. Sending patches for review
+
+    git send-email <commit list|directory>
+
+  will send the patches created by git format-patch or directly generates
+  them. All the email fields can be configured in the global/local
+  configuration or overridden by command line.
+  Note that this tool must often be installed separately (e.g. git-email
+  package on Debian-based distros).
+
+12. Pushing changes to remote trees
+
+    git push
+
+  Will push the changes to the default remote (origin).
+  Git will prevent you from pushing changes if the local and remote trees are
+  out of sync. Refer to 2 and 2.a to sync the local tree.
+
+    git remote add <name> <url>
+
+  Will add additional remote with a name reference, it is useful if you want
+  to push your local branch for review on a remote host.
+
+    git push <remote> <refspec>
+
+  Will push the changes to the remote repository. Omitting refspec makes git
+  push update all the remote branches matching the local ones.
+
+13. Finding a specific svn revision
+
+  Since version 1.7.1 git supports ':/foo' syntax for specifying commits
+  based on a regular expression. see man gitrevisions
+
+    git show :/'as revision 23456'
+
+  will show the svn changeset r23456. With older git versions searching in
+  the git log output is the easiest option (especially if a pager with
+  search capabilities is used).
+  This commit can be checked out with
+
+    git checkout -b svn_23456 :/'as revision 23456'
+
+  or for git < 1.7.1 with
+
+    git checkout -b svn_23456 $SHA1
+
+  where $SHA1 is the commit SHA1 from the 'git log' output.
+
+
+Contact the project admins <root at ffmpeg dot org> if you have technical
+problems with the GIT server.

doc/indevs.texi

@@ -1,7 +1,7 @@
 @chapter Input Devices
 @c man begin INPUT DEVICES
 
-Input devices are configured elements in FFmpeg which enable accessing
+Input devices are configured elements in FFmpeg which allow to access
 the data coming from a multimedia device attached to your system.
 
 When you configure your FFmpeg build, all the supported input devices
@@ -13,8 +13,8 @@ You can disable all the input devices using the configure option
 option "--enable-indev=@var{INDEV}", or you can disable a particular
 input device using the option "--disable-indev=@var{INDEV}".
 
-The option "-devices" of the ff* tools will display the list of
-supported input devices.
+The option "-formats" of the ff* tools will display the list of
+supported input devices (amongst the demuxers).
 
 A description of the currently available input devices follows.
 
@@ -51,180 +51,10 @@ ffmpeg -f alsa -i hw:0 alsaout.wav
 For more information see:
 @url{http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html}
 
-@section avfoundation
-
-AVFoundation input device.
-
-AVFoundation is the currently recommended framework by Apple for streamgrabbing on OSX >= 10.7 as well as on iOS.
-The older QTKit framework has been marked deprecated since OSX version 10.7.
-
-The input filename has to be given in the following syntax:
-@example
--i "[[VIDEO]:[AUDIO]]"
-@end example
-The first entry selects the video input while the latter selects the audio input.
-The stream has to be specified by the device name or the device index as shown by the device list.
-Alternatively, the video and/or audio input device can be chosen by index using the
-@option{
-    -video_device_index <INDEX>
-}
-and/or
-@option{
-    -audio_device_index <INDEX>
-}
-, overriding any
-device name or index given in the input filename.
-
-All available devices can be enumerated by using @option{-list_devices true}, listing
-all device names and corresponding indices.
-
-There are two device name aliases:
-@table @code
-
-@item default
-Select the AVFoundation default device of the corresponding type.
-
-@item none
-Do not record the corresponding media type.
-This is equivalent to specifying an empty device name or index.
-
-@end table
-
-@subsection Options
-
-AVFoundation supports the following options:
-
-@table @option
-
-@item -list_devices <TRUE|FALSE>
-If set to true, a list of all available input devices is given showing all
-device names and indices.
-
-@item -video_device_index <INDEX>
-Specify the video device by its index. Overrides anything given in the input filename.
-
-@item -audio_device_index <INDEX>
-Specify the audio device by its index. Overrides anything given in the input filename.
-
-@item -pixel_format <FORMAT>
-Request the video device to use a specific pixel format.
-If the specified format is not supported, a list of available formats is given
-und the first one in this list is used instead. Available pixel formats are:
-@code{monob, rgb555be, rgb555le, rgb565be, rgb565le, rgb24, bgr24, 0rgb, bgr0, 0bgr, rgb0,
- bgr48be, uyvy422, yuva444p, yuva444p16le, yuv444p, yuv422p16, yuv422p10, yuv444p10,
- yuv420p, nv12, yuyv422, gray}
-
-@end table
-
-@subsection Examples
-
-@itemize
-
-@item
-Print the list of AVFoundation supported devices and exit:
-@example
-$ ffmpeg -f avfoundation -list_devices true -i ""
-@end example
-
-@item
-Record video from video device 0 and audio from audio device 0 into out.avi:
-@example
-$ ffmpeg -f avfoundation -i "0:0" out.avi
-@end example
-
-@item
-Record video from video device 2 and audio from audio device 1 into out.avi:
-@example
-$ ffmpeg -f avfoundation -video_device_index 2 -i ":1" out.avi
-@end example
-
-@item
-Record video from the system default video device using the pixel format bgr0 and do not record any audio into out.avi:
-@example
-$ ffmpeg -f avfoundation -pixel_format bgr0 -i "default:none" out.avi
-@end example
-
-@end itemize
-
 @section bktr
 
 BSD video input device.
 
-@section decklink
-
-The decklink input device provides capture capabilities for Blackmagic
-DeckLink devices.
-
-To enable this input device, you need the Blackmagic DeckLink SDK and you
-need to configure with the appropriate @code{--extra-cflags}
-and @code{--extra-ldflags}.
-On Windows, you need to run the IDL files through @command{widl}.
-
-DeckLink is very picky about the formats it supports. Pixel format is
-uyvy422 or v210, framerate and video size must be determined for your device with
-@command{-list_formats 1}. Audio sample rate is always 48 kHz and the number
-of channels can be 2, 8 or 16.
-
-@subsection Options
-
-@table @option
-
-@item list_devices
-If set to @option{true}, print a list of devices and exit.
-Defaults to @option{false}.
-
-@item list_formats
-If set to @option{true}, print a list of supported formats and exit.
-Defaults to @option{false}.
-
-@item bm_v210
-If set to @samp{1}, video is captured in 10 bit v210 instead
-of uyvy422. Not all Blackmagic devices support this option.
-
-@end table
-
-@subsection Examples
-
-@itemize
-
-@item
-List input devices:
-@example
-ffmpeg -f decklink -list_devices 1 -i dummy
-@end example
-
-@item
-List supported formats:
-@example
-ffmpeg -f decklink -list_formats 1 -i 'Intensity Pro'
-@end example
-
-@item
-Capture video clip at 1080i50 (format 11):
-@example
-ffmpeg -f decklink -i 'Intensity Pro@@11' -acodec copy -vcodec copy output.avi
-@end example
-
-@item
-Capture video clip at 1080i50 10 bit:
-@example
-ffmpeg -bm_v210 1 -f decklink -i 'UltraStudio Mini Recorder@@11' -acodec copy -vcodec copy output.avi
-@end example
-
-@item
-Capture video clip at 720p50 with 32bit audio:
-@example
-ffmpeg -bm_audiodepth 32 -f decklink -i 'UltraStudio Mini Recorder@@14' -acodec copy -vcodec copy output.avi
-@end example
-
-@item
-Capture video clip at 576i50 with 8 audio channels:
-@example
-ffmpeg -bm_channels 8 -f decklink -i 'UltraStudio Mini Recorder@@3' -acodec copy -vcodec copy output.avi
-@end example
-
-@end itemize
-
 @section dshow
 
 Windows DirectShow input device.
@@ -242,7 +72,7 @@ The input name should be in the format:
 @end example
 
 where @var{TYPE} can be either @var{audio} or @var{video},
-and @var{NAME} is the device's name or alternative name..
+and @var{NAME} is the device's name.
 
 @subsection Options
 
@@ -295,61 +125,6 @@ Setting this value too low can degrade performance.
 See also
 @url{http://msdn.microsoft.com/en-us/library/windows/desktop/dd377582(v=vs.85).aspx}
 
-@item video_pin_name
-Select video capture pin to use by name or alternative name.
-
-@item audio_pin_name
-Select audio capture pin to use by name or alternative name.
-
-@item crossbar_video_input_pin_number
-Select video input pin number for crossbar device. This will be
-routed to the crossbar device's Video Decoder output pin.
-Note that changing this value can affect future invocations
-(sets a new default) until system reboot occurs.
-
-@item crossbar_audio_input_pin_number
-Select audio input pin number for crossbar device. This will be
-routed to the crossbar device's Audio Decoder output pin.
-Note that changing this value can affect future invocations
-(sets a new default) until system reboot occurs.
-
-@item show_video_device_dialog
-If set to @option{true}, before capture starts, popup a display dialog
-to the end user, allowing them to change video filter properties
-and configurations manually.
-Note that for crossbar devices, adjusting values in this dialog
-may be needed at times to toggle between PAL (25 fps) and NTSC (29.97)
-input frame rates, sizes, interlacing, etc.  Changing these values can
-enable different scan rates/frame rates and avoiding green bars at
-the bottom, flickering scan lines, etc.
-Note that with some devices, changing these properties can also affect future
-invocations (sets new defaults) until system reboot occurs.
-
-@item show_audio_device_dialog
-If set to @option{true}, before capture starts, popup a display dialog
-to the end user, allowing them to change audio filter properties
-and configurations manually.
-
-@item show_video_crossbar_connection_dialog
-If set to @option{true}, before capture starts, popup a display
-dialog to the end user, allowing them to manually
-modify crossbar pin routings, when it opens a video device.
-
-@item show_audio_crossbar_connection_dialog
-If set to @option{true}, before capture starts, popup a display
-dialog to the end user, allowing them to manually
-modify crossbar pin routings, when it opens an audio device.
-
-@item show_analog_tv_tuner_dialog
-If set to @option{true}, before capture starts, popup a display
-dialog to the end user, allowing them to manually
-modify TV channels and frequencies.
-
-@item show_analog_tv_tuner_audio_dialog
-If set to @option{true}, before capture starts, popup a display
-dialog to the end user, allowing them to manually
-modify TV audio (like mono vs. stereo, Language A,B or C).
-
 @end table
 
 @subsection Examples
@@ -386,19 +161,6 @@ Print the list of supported options in selected device and exit:
 $ ffmpeg -list_options true -f dshow -i video="Camera"
 @end example
 
-@item
-Specify pin names to capture by name or alternative name, specify alternative device name:
-@example
-$ ffmpeg -f dshow -audio_pin_name "Audio Out" -video_pin_name 2 -i video=video="@@device_pnp_\\?\pci#ven_1a0a&dev_6200&subsys_62021461&rev_01#4&e2c7dd6&0&00e1#@{65e8773d-8f56-11d0-a3b9-00a0c9223196@}\@{ca465100-deb0-4d59-818f-8c477184adf6@}":audio="Microphone"
-@end example
-
-@item
-Configure a crossbar device, specifying crossbar pins, allow user to adjust video capture properties at startup:
-@example
-$ ffmpeg -f dshow -show_video_device_dialog true -crossbar_video_input_pin_number 0
-     -crossbar_audio_input_pin_number 3 -i video="AVerMedia BDA Analog Capture":audio="AVerMedia BDA Analog Capture"
-@end example
-
 @end itemize
 
 @section dv1394
@@ -430,81 +192,6 @@ ffmpeg -f fbdev -frames:v 1 -r 1 -i /dev/fb0 screenshot.jpeg
 
 See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1).
 
-@section gdigrab
-
-Win32 GDI-based screen capture device.
-
-This device allows you to capture a region of the display on Windows.
-
-There are two options for the input filename:
-@example
-desktop
-@end example
-or
-@example
-title=@var{window_title}
-@end example
-
-The first option will capture the entire desktop, or a fixed region of the
-desktop. The second option will instead capture the contents of a single
-window, regardless of its position on the screen.
-
-For example, to grab the entire desktop using @command{ffmpeg}:
-@example
-ffmpeg -f gdigrab -framerate 6 -i desktop out.mpg
-@end example
-
-Grab a 640x480 region at position @code{10,20}:
-@example
-ffmpeg -f gdigrab -framerate 6 -offset_x 10 -offset_y 20 -video_size vga -i desktop out.mpg
-@end example
-
-Grab the contents of the window named "Calculator"
-@example
-ffmpeg -f gdigrab -framerate 6 -i title=Calculator out.mpg
-@end example
-
-@subsection Options
-
-@table @option
-@item draw_mouse
-Specify whether to draw the mouse pointer. Use the value @code{0} to
-not draw the pointer. Default value is @code{1}.
-
-@item framerate
-Set the grabbing frame rate. Default value is @code{ntsc},
-corresponding to a frame rate of @code{30000/1001}.
-
-@item show_region
-Show grabbed region on screen.
-
-If @var{show_region} is specified with @code{1}, then the grabbing
-region will be indicated on screen. With this option, it is easy to
-know what is being grabbed if only a portion of the screen is grabbed.
-
-Note that @var{show_region} is incompatible with grabbing the contents
-of a single window.
-
-For example:
-@example
-ffmpeg -f gdigrab -show_region 1 -framerate 6 -video_size cif -offset_x 10 -offset_y 20 -i desktop out.mpg
-@end example
-
-@item video_size
-Set the video frame size. The default is to capture the full screen if @file{desktop} is selected, or the full window size if @file{title=@var{window_title}} is selected.
-
-@item offset_x
-When capturing a region with @var{video_size}, set the distance from the left edge of the screen or desktop.
-
-Note that the offset calculation is from the top left corner of the primary monitor on Windows. If you have a monitor positioned to the left of your primary monitor, you will need to use a negative @var{offset_x} value to move the region to that monitor.
-
-@item offset_y
-When capturing a region with @var{video_size}, set the distance from the top edge of the screen or desktop.
-
-Note that the offset calculation is from the top left corner of the primary monitor on Windows. If you have a monitor positioned above your primary monitor, you will need to use a negative @var{offset_y} value to move the region to that monitor.
-
-@end table
-
 @section iec61883
 
 FireWire DV/HDV input device using libiec61883.
@@ -533,7 +220,7 @@ not work and result in undefined behavior.
 The values @option{auto}, @option{dv} and @option{hdv} are supported.
 
 @item dvbuffer
-Set maximum size of buffer for incoming data, in frames. For DV, this
+Set maxiumum size of buffer for incoming data, in frames. For DV, this
 is an exact value. For HDV, it is not frame exact, since HDV does
 not have a fixed frame size.
 
@@ -638,14 +325,6 @@ generated by the device.
 The first unlabelled output is automatically assigned to the "out0"
 label, but all the others need to be specified explicitly.
 
-The suffix "+subcc" can be appended to the output label to create an extra
-stream with the closed captions packets attached to that output
-(experimental; only for EIA-608 / CEA-708 for now).
-The subcc streams are created after all the normal streams, in the order of
-the corresponding stream.
-For example, if there is "out19+subcc", "out7+subcc" and up to "out42", the
-stream #43 is subcc for stream #7 and stream #44 is subcc for stream #19.
-
 If not specified defaults to the filename specified for the input
 device.
 
@@ -692,63 +371,12 @@ Read an audio stream and a video stream and play it back with
 ffplay -f lavfi "movie=test.avi[out0];amovie=test.wav[out1]"
 @end example
 
-@item
-Dump decoded frames to images and closed captions to a file (experimental):
-@example
-ffmpeg -f lavfi -i "movie=test.ts[out0+subcc]" -map v frame%08d.png -map s -c copy -f rawvideo subcc.bin
-@end example
-
 @end itemize
 
-@section libcdio
-
-Audio-CD input device based on libcdio.
-
-To enable this input device during configuration you need libcdio
-installed on your system. It requires the configure option
-@code{--enable-libcdio}.
-
-This device allows playing and grabbing from an Audio-CD.
-
-For example to copy with @command{ffmpeg} the entire Audio-CD in @file{/dev/sr0},
-you may run the command:
-@example
-ffmpeg -f libcdio -i /dev/sr0 cd.wav
-@end example
-
-@subsection Options
-@table @option
-@item speed
-Set drive reading speed. Default value is 0.
-
-The speed is specified CD-ROM speed units. The speed is set through
-the libcdio @code{cdio_cddap_speed_set} function. On many CD-ROM
-drives, specifying a value too large will result in using the fastest
-speed.
-
-@item paranoia_mode
-Set paranoia recovery mode flags. It accepts one of the following values:
-
-@table @samp
-@item disable
-@item verify
-@item overlap
-@item neverskip
-@item full
-@end table
-
-Default value is @samp{disable}.
-
-For more information about the available recovery modes, consult the
-paranoia project documentation.
-@end table
-
 @section libdc1394
 
 IIDC1394 input device, based on libdc1394 and libraw1394.
 
-Requires the configure option @code{--enable-libdc1394}.
-
 @section openal
 
 The OpenAL input device provides audio capture on all systems with a
@@ -781,7 +409,7 @@ OpenAL is part of Core Audio, the official Mac OS X Audio interface.
 See @url{http://developer.apple.com/technologies/mac/audio-and-video.html}
 @end table
 
-This device allows one to capture from an audio input device handled
+This device allows to capture from an audio input device handled
 through OpenAL.
 
 You need to specify the name of the device to capture in the provided
@@ -903,33 +531,6 @@ Record a stream from default device:
 ffmpeg -f pulse -i default /tmp/pulse.wav
 @end example
 
-@section qtkit
-
-QTKit input device.
-
-The filename passed as input is parsed to contain either a device name or index.
-The device index can also be given by using -video_device_index.
-A given device index will override any given device name.
-If the desired device consists of numbers only, use -video_device_index to identify it.
-The default device will be chosen if an empty string  or the device name "default" is given.
-The available devices can be enumerated by using -list_devices.
-
-@example
-ffmpeg -f qtkit -i "0" out.mpg
-@end example
-
-@example
-ffmpeg -f qtkit -video_device_index 0 -i "" out.mpg
-@end example
-
-@example
-ffmpeg -f qtkit -i "default" out.mpg
-@end example
-
-@example
-ffmpeg -f qtkit -list_devices true -i ""
-@end example
-
 @section sndio
 
 sndio input device.
@@ -1016,7 +617,7 @@ Select the pixel format (only valid for raw video input).
 
 @item input_format
 Set the preferred pixel format (for raw video) or a codec name.
-This option allows one to select the input format, when several are
+This option allows to select the input format, when several are
 available.
 
 @item framerate
@@ -1077,14 +678,7 @@ other filename will be interpreted as device number 0.
 
 X11 video input device.
 
-To enable this input device during configuration you need libxcb
-installed on your system. It will be automatically detected during
-configuration.
-
-Alternatively, the configure option @option{--enable-x11grab} exists
-for legacy Xlib users.
-
-This device allows one to capture a region of an X11 display.
+This device allows to capture a region of an X11 display.
 
 The filename passed as input has the syntax:
 @example
@@ -1100,12 +694,10 @@ omitted, and defaults to "localhost". The environment variable
 area with respect to the top-left border of the X11 screen. They
 default to 0.
 
-Check the X11 documentation (e.g. @command{man X}) for more detailed
-information.
+Check the X11 documentation (e.g. man X) for more detailed information.
 
-Use the @command{xdpyinfo} program for getting basic information about
-the properties of your X11 display (e.g. grep for "name" or
-"dimensions").
+Use the @command{dpyinfo} program for getting basic information about the
+properties of your X11 display (e.g. grep for "name" or "dimensions").
 
 For example to grab from @file{:0.0} using @command{ffmpeg}:
 @example
@@ -1154,10 +746,6 @@ If @var{show_region} is specified with @code{1}, then the grabbing
 region will be indicated on screen. With this option, it is easy to
 know what is being grabbed if only a portion of the screen is grabbed.
 
-@item region_border
-Set the region border thickness if @option{-show_region 1} is used.
-Range is 1 to 128 and default is 3 (XCB-based x11grab only).
-
 For example:
 @example
 ffmpeg -f x11grab -show_region 1 -framerate 25 -video_size cif -i :0.0+10,20 out.mpg
@@ -1170,22 +758,6 @@ ffmpeg -f x11grab -follow_mouse centered -show_region 1 -framerate 25 -video_siz
 
 @item video_size
 Set the video frame size. Default value is @code{vga}.
-
-@item use_shm
-Use the MIT-SHM extension for shared memory. Default value is @code{1}.
-It may be necessary to disable it for remote displays (legacy x11grab
-only).
 @end table
 
-@subsection @var{grab_x} @var{grab_y} AVOption
-
-The syntax is:
-@example
--grab_x @var{x_offset} -grab_y @var{y_offset}
-@end example
-
-Set the grabbing region coordinates. They are expressed as offset from the top left
-corner of the X11 window. The default value is 0.
-
-
 @c man end INPUT DEVICES

doc/issue_tracker.txt

@@ -22,7 +22,7 @@ a mail for every change to every issue.
 (the above does all work already after light testing)
 
 The subscription URL for the ffmpeg-trac list is:
-http(s)://lists.ffmpeg.org/mailman/listinfo/ffmpeg-trac
+http(s)://ffmpeg.org/mailman/listinfo/ffmpeg-trac
 The URL of the webinterface of the tracker is:
 http(s)://trac.ffmpeg.org
 

doc/libavcodec.texi

@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8
 
 @settitle Libavcodec Documentation
 @titlepage

doc/libavdevice.texi

@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8
 
 @settitle Libavdevice Documentation
 @titlepage

doc/libavfilter.texi

@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8
 
 @settitle Libavfilter Documentation
 @titlepage

doc/libavformat.texi

@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8
 
 @settitle Libavformat Documentation
 @titlepage

doc/libavutil.texi

@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8
 
 @settitle Libavutil Documentation
 @titlepage
@@ -17,25 +16,7 @@ The libavutil library is a utility library to aid portable
 multimedia programming. It contains safe portable string functions,
 random number generators, data structures, additional mathematics
 functions, cryptography and multimedia related functionality (like
-enumerations for pixel and sample formats). It is not a library for
-code needed by both libavcodec and libavformat.
-
-The goals for this library is to be:
-
-@table @strong
-@item Modular
-It should have few interdependencies and the possibility of disabling individual
-parts during @command{./configure}.
-
-@item Small
-Both sources and objects should be small.
-
-@item Efficient
-It should have low CPU and memory usage.
-
-@item Useful
-It should avoid useless features that almost no one needs.
-@end table
+enumerations for pixel and sample formats).
 
 @c man end DESCRIPTION
 

doc/libswresample.texi

@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8
 
 @settitle Libswresample Documentation
 @titlepage

doc/libswscale.texi

@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8
 
 @settitle Libswscale Documentation
 @titlepage

doc/muxers.texi

@@ -23,8 +23,6 @@ A description of some of the currently available muxers follows.
 
 Audio Interchange File Format muxer.
 
-@subsection Options
-
 It accepts the following options:
 
 @table @option
@@ -51,10 +49,6 @@ The output of the muxer consists of a single line of the form:
 CRC=0x@var{CRC}, where @var{CRC} is a hexadecimal number 0-padded to
 8 digits containing the CRC for all the decoded input frames.
 
-See also the @ref{framecrc} muxer.
-
-@subsection Examples
-
 For example to compute the CRC of the input, and store it in the file
 @file{out.crc}:
 @example
@@ -74,6 +68,8 @@ and the input video converted to MPEG-2 video, use the command:
 ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f crc -
 @end example
 
+See also the @ref{framecrc} muxer.
+
 @anchor{framecrc}
 @section framecrc
 
@@ -93,8 +89,6 @@ packet of the form:
 @var{CRC} is a hexadecimal number 0-padded to 8 digits containing the
 CRC of the packet.
 
-@subsection Examples
-
 For example to compute the CRC of the audio and video frames in
 @file{INPUT}, converted to raw audio and video packets, and store it
 in the file @file{out.crc}:
@@ -138,8 +132,6 @@ packet of the form:
 @var{MD5} is a hexadecimal number representing the computed MD5 hash
 for the packet.
 
-@subsection Examples
-
 For example to compute the MD5 of the audio and video frames in
 @file{INPUT}, converted to raw audio and video packets, and store it
 in the file @file{out.md5}:
@@ -154,129 +146,30 @@ ffmpeg -i INPUT -f framemd5 -
 
 See also the @ref{md5} muxer.
 
-@anchor{gif}
-@section gif
-
-Animated GIF muxer.
-
-It accepts the following options:
-
-@table @option
-@item loop
-Set the number of times to loop the output. Use @code{-1} for no loop, @code{0}
-for looping indefinitely (default).
-
-@item final_delay
-Force the delay (expressed in centiseconds) after the last frame. Each frame
-ends with a delay until the next frame. The default is @code{-1}, which is a
-special value to tell the muxer to re-use the previous delay. In case of a
-loop, you might want to customize this value to mark a pause for instance.
-@end table
-
-For example, to encode a gif looping 10 times, with a 5 seconds delay between
-the loops:
-@example
-ffmpeg -i INPUT -loop 10 -final_delay 500 out.gif
-@end example
-
-Note 1: if you wish to extract the frames in separate GIF files, you need to
-force the @ref{image2} muxer:
-@example
-ffmpeg -i INPUT -c:v gif -f image2 "out%d.gif"
-@end example
-
-Note 2: the GIF format has a very small time base: the delay between two frames
-can not be smaller than one centi second.
-
 @anchor{hls}
 @section hls
 
 Apple HTTP Live Streaming muxer that segments MPEG-TS according to
-the HTTP Live Streaming (HLS) specification.
+the HTTP Live Streaming specification.
 
-It creates a playlist file, and one or more segment files. The output filename
-specifies the playlist filename.
+It creates a playlist file and numbered segment files. The output
+filename specifies the playlist filename; the segment filenames
+receive the same basename as the playlist, a sequential number and
+a .ts extension.
 
-By default, the muxer creates a file for each segment produced. These files
-have the same name as the playlist, followed by a sequential number and a
-.ts extension.
-
-For example, to convert an input file with @command{ffmpeg}:
 @example
 ffmpeg -i in.nut out.m3u8
 @end example
-This example will produce the playlist, @file{out.m3u8}, and segment files:
-@file{out0.ts}, @file{out1.ts}, @file{out2.ts}, etc.
-
-See also the @ref{segment} muxer, which provides a more generic and
-flexible implementation of a segmenter, and can be used to perform HLS
-segmentation.
-
-@subsection Options
-
-This muxer supports the following options:
 
 @table @option
-@item hls_time @var{seconds}
-Set the segment length in seconds. Default value is 2.
-
-@item hls_list_size @var{size}
-Set the maximum number of playlist entries. If set to 0 the list file
-will contain all the segments. Default value is 5.
-
-@item hls_ts_options @var{options_list}
-Set output format options using a :-separated list of key=value
-parameters. Values containing @code{:} special characters must be
-escaped.
-
-@item hls_wrap @var{wrap}
-Set the number after which the segment filename number (the number
-specified in each segment file) wraps. If set to 0 the number will be
-never wrapped. Default value is 0.
-
-This option is useful to avoid to fill the disk with many segment
-files, and limits the maximum number of segment files written to disk
-to @var{wrap}.
-
-@item start_number @var{number}
-Start the playlist sequence number from @var{number}. Default value is
-0.
-
-@item hls_allow_cache @var{allowcache}
-Explicitly set whether the client MAY (1) or MUST NOT (0) cache media segments.
-
-@item hls_base_url @var{baseurl}
-Append @var{baseurl} to every entry in the playlist.
-Useful to generate playlists with absolute paths.
-
-Note that the playlist sequence number must be unique for each segment
-and it is not to be confused with the segment filename sequence number
-which can be cyclic, for example if the @option{wrap} option is
-specified.
-
-@item hls_segment_filename @var{filename}
-Set the segment filename. Unless hls_flags single_file is set @var{filename}
-is used as a string format with the segment number:
-@example
-ffmpeg in.nut -hls_segment_filename 'file%03d.ts' out.m3u8
-@end example
-This example will produce the playlist, @file{out.m3u8}, and segment files:
-@file{file000.ts}, @file{file001.ts}, @file{file002.ts}, etc.
-
-@item hls_flags single_file
-If this flag is set, the muxer will store all segments in a single MPEG-TS
-file, and will use byte ranges in the playlist. HLS playlists generated with
-this way will have the version number 4.
-For example:
-@example
-ffmpeg -i in.nut -hls_flags single_file out.m3u8
-@end example
-Will produce the playlist, @file{out.m3u8}, and a single segment file,
-@file{out.ts}.
-
-@item hls_flags delete_segments
-Segment files removed from the playlist are deleted after a period of time
-equal to the duration of the segment plus the duration of the playlist.
+@item -hls_time @var{seconds}
+Set the segment length in seconds.
+@item -hls_list_size @var{size}
+Set the maximum number of playlist entries.
+@item -hls_wrap @var{wrap}
+Set the number after which index wraps.
+@item -start_number @var{number}
+Start the sequence from @var{number}.
 @end table
 
 @anchor{ico}
@@ -342,8 +235,6 @@ The pattern "img%%-%d.jpg" will specify a sequence of filenames of the
 form @file{img%-1.jpg}, @file{img%-2.jpg}, ..., @file{img%-10.jpg},
 etc.
 
-@subsection Examples
-
 The following example shows how to use @command{ffmpeg} for creating a
 sequence of files @file{img-001.jpeg}, @file{img-002.jpeg}, ...,
 taking one image every second from the input video:
@@ -366,31 +257,16 @@ Note also that the pattern must not necessarily contain "%d" or
 ffmpeg -i in.avi -f image2 -frames:v 1 img.jpeg
 @end example
 
-The @option{strftime} option allows you to expand the filename with
-date and time information. Check the documentation of
-the @code{strftime()} function for the syntax.
-
-For example to generate image files from the @code{strftime()}
-"%Y-%m-%d_%H-%M-%S" pattern, the following @command{ffmpeg} command
-can be used:
-@example
-ffmpeg -f v4l2 -r 1 -i /dev/video0 -f image2 -strftime 1 "%Y-%m-%d_%H-%M-%S.jpg"
-@end example
-
-@subsection Options
-
 @table @option
-@item start_number
-Start the sequence from the specified number. Default value is 0.
+@item start_number @var{number}
+Start the sequence from @var{number}. Default value is 1. Must be a
+non-negative number.
 
-@item update
-If set to 1, the filename will always be interpreted as just a
-filename, not a pattern, and the corresponding file will be continuously
-overwritten with new images. Default value is 0.
+@item -update @var{number}
+If @var{number} is nonzero, the filename will always be interpreted as just a
+filename, not a pattern, and this file will be continuously overwritten with new
+images.
 
-@item strftime
-If set to 1, expand the filename with date and time information from
-@code{strftime()}. Default value is 0.
 @end table
 
 The image muxer supports the .Y.U.V image file format. This format is
@@ -405,27 +281,25 @@ Matroska container muxer.
 
 This muxer implements the matroska and webm container specs.
 
-@subsection Metadata
-
 The recognized metadata settings in this muxer are:
 
 @table @option
-@item title
-Set title name provided to a single track.
 
-@item language
-Specify the language of the track in the Matroska languages form.
+@item title=@var{title name}
+Name provided to a single track
+@end table
 
-The language can be either the 3 letters bibliographic ISO-639-2 (ISO
-639-2/B) form (like "fre" for French), or a language code mixed with a
-country code for specialities in languages (like "fre-ca" for Canadian
-French).
+@table @option
 
-@item stereo_mode
-Set stereo 3D video layout of two views in a single video track.
+@item language=@var{language name}
+Specifies the language of the track in the Matroska languages form
+@end table
 
-The following values are recognized:
-@table @samp
+@table @option
+
+@item stereo_mode=@var{mode}
+Stereo 3D video layout of two views in a single video track
+@table @option
 @item mono
 video is not stereo
 @item left_right
@@ -464,11 +338,10 @@ For example a 3D WebM clip can be created using the following command line:
 ffmpeg -i sample_left_right_clip.mpg -an -c:v libvpx -metadata stereo_mode=left_right -y stereo_clip.webm
 @end example
 
-@subsection Options
-
 This muxer supports the following options:
 
 @table @option
+
 @item reserve_index_space
 By default, this muxer writes the index for seeking (called cues in Matroska
 terms) at the end of the file, because it cannot know in advance how much space
@@ -483,6 +356,7 @@ for most use cases should be about 50kB per hour of video.
 
 Note that cues are only written if the output is seekable and this option will
 have no effect if it is not.
+
 @end table
 
 @anchor{md5}
@@ -512,9 +386,7 @@ ffmpeg -i INPUT -f md5 -
 
 See also the @ref{framemd5} muxer.
 
-@section mov, mp4, ismv
-
-MOV/MP4/ISMV (Smooth Streaming) muxer.
+@section MOV/MP4/ISMV
 
 The mov/mp4/ismv muxer supports fragmentation. Normally, a MOV/MP4
 file has all the metadata about all packets stored in one location
@@ -530,8 +402,6 @@ very long files (since writing normal MOV/MP4 files stores info about
 every single packet in memory until the file is closed). The downside
 is that it is less compatible with other applications.
 
-@subsection Options
-
 Fragmentation is enabled by setting one of the AVOptions that define
 how to cut the file into fragments:
 
@@ -571,6 +441,7 @@ a short portion of the file. With this option set, there is no initial
 mdat atom, and the moov atom only describes the tracks but has
 a zero duration.
 
+Files written with this option set do not work in QuickTime.
 This option is implicitly set when writing ismv (Smooth Streaming) files.
 @item -movflags separate_moof
 Write a separate moof (movie fragment) atom for each track. Normally,
@@ -585,26 +456,8 @@ This operation can take a while, and will not work in various situations such
 as fragmented output, thus it is not enabled by default.
 @item -movflags rtphint
 Add RTP hinting tracks to the output file.
-@item -movflags disable_chpl
-Disable Nero chapter markers (chpl atom).  Normally, both Nero chapters
-and a QuickTime chapter track are written to the file. With this option
-set, only the QuickTime chapter track will be written. Nero chapters can
-cause failures when the file is reprocessed with certain tagging programs, like
-mp3Tag 2.61a and iTunes 11.3, most likely other versions are affected as well.
-@item -movflags omit_tfhd_offset
-Do not write any absolute base_data_offset in tfhd atoms. This avoids
-tying fragments to absolute byte positions in the file/streams.
-@item -movflags default_base_moof
-Similarly to the omit_tfhd_offset, this flag avoids writing the
-absolute base_data_offset field in tfhd atoms, but does so by using
-the new default-base-is-moof flag instead. This flag is new from
-14496-12:2012. This may make the fragments easier to parse in certain
-circumstances (avoiding basing track fragment location calculations
-on the implicit end of the previous track fragment).
 @end table
 
-@subsection Example
-
 Smooth Streaming content can be pushed in real time to a publishing
 point on IIS with this muxer. Example:
 @example
@@ -613,38 +466,26 @@ ffmpeg -re @var{<normal input/transcoding options>} -movflags isml+frag_keyframe
 
 @section mp3
 
-The MP3 muxer writes a raw MP3 stream with the following optional features:
-@itemize @bullet
-@item
-An ID3v2 metadata header at the beginning (enabled by default). Versions 2.3 and
-2.4 are supported, the @code{id3v2_version} private option controls which one is
-used (3 or 4). Setting @code{id3v2_version} to 0 disables the ID3v2 header
-completely.
+The MP3 muxer writes a raw MP3 stream with an ID3v2 header at the beginning and
+optionally an ID3v1 tag at the end. ID3v2.3 and ID3v2.4 are supported, the
+@code{id3v2_version} option controls which one is used. The legacy ID3v1 tag is
+not written by default, but may be enabled with the @code{write_id3v1} option.
 
-The muxer supports writing attached pictures (APIC frames) to the ID3v2 header.
-The pictures are supplied to the muxer in form of a video stream with a single
-packet. There can be any number of those streams, each will correspond to a
-single APIC frame.  The stream metadata tags @var{title} and @var{comment} map
-to APIC @var{description} and @var{picture type} respectively. See
+For seekable output the muxer also writes a Xing frame at the beginning, which
+contains the number of frames in the file. It is useful for computing duration
+of VBR files.
+
+The muxer supports writing ID3v2 attached pictures (APIC frames). The pictures
+are supplied to the muxer in form of a video stream with a single packet. There
+can be any number of those streams, each will correspond to a single APIC frame.
+The stream metadata tags @var{title} and @var{comment} map to APIC
+@var{description} and @var{picture type} respectively. See
 @url{http://id3.org/id3v2.4.0-frames} for allowed picture types.
 
 Note that the APIC frames must be written at the beginning, so the muxer will
 buffer the audio frames until it gets all the pictures. It is therefore advised
 to provide the pictures as soon as possible to avoid excessive buffering.
 
-@item
-A Xing/LAME frame right after the ID3v2 header (if present). It is enabled by
-default, but will be written only if the output is seekable. The
-@code{write_xing} private option can be used to disable it.  The frame contains
-various information that may be useful to the decoder, like the audio duration
-or encoder delay.
-
-@item
-A legacy ID3v1 tag at the end of the file (disabled by default). It may be
-enabled with the @code{write_id3v1} private option, but as its capabilities are
-very limited, its usage is not recommended.
-@end itemize
-
 Examples:
 
 Write an mp3 with an ID3v2.3 header and an ID3v1 footer:
@@ -659,24 +500,12 @@ ffmpeg -i input.mp3 -i cover.png -c copy -map 0 -map 1
 -metadata:s:v title="Album cover" -metadata:s:v comment="Cover (Front)" out.mp3
 @end example
 
-Write a "clean" MP3 without any extra features:
-@example
-ffmpeg -i input.wav -write_xing 0 -id3v2_version 0 out.mp3
-@end example
-
 @section mpegts
 
 MPEG transport stream muxer.
 
 This muxer implements ISO 13818-1 and part of ETSI EN 300 468.
 
-The recognized metadata settings in mpegts muxer are @code{service_provider}
-and @code{service_name}. If they are not set the default for
-@code{service_provider} is "FFmpeg" and the default for
-@code{service_name} is "Service01".
-
-@subsection Options
-
 The muxer options are:
 
 @table @option
@@ -689,9 +518,6 @@ Set the transport_stream_id (default 0x0001). This identifies a
 transponder in DVB.
 @item -mpegts_service_id @var{number}
 Set the service_id (default 0x0001) also known as program in DVB.
-@item -mpegts_service_type @var{number}
-Set the program service_type (default @var{digital_tv}), see below
-a list of pre defined values.
 @item -mpegts_pmt_start_pid @var{number}
 Set the first PID for PMT (default 0x1000, max 0x1f00).
 @item -mpegts_start_pid @var{number}
@@ -699,10 +525,7 @@ Set the first PID for data packets (default 0x0100, max 0x0f00).
 @item -mpegts_m2ts_mode @var{number}
 Enable m2ts mode if set to 1. Default value is -1 which disables m2ts mode.
 @item -muxrate @var{number}
-Set a constant muxrate (default VBR).
-@item -pcr_period @var{numer}
-Override the default PCR retransmission time (default 20ms), ignored
-if variable muxrate is selected.
+Set muxrate.
 @item -pes_payload_size @var{number}
 Set minimum PES packet payload in bytes.
 @item -mpegts_flags @var{flags}
@@ -726,27 +549,6 @@ ffmpeg -i source2.ts -codec copy -f mpegts -tables_version 1 udp://1.1.1.1:1111
 @end example
 @end table
 
-Option mpegts_service_type accepts the following values:
-
-@table @option
-@item hex_value
-Any hexdecimal value between 0x01 to 0xff as defined in ETSI 300 468.
-@item digital_tv
-Digital TV service.
-@item digital_radio
-Digital Radio service.
-@item teletext
-Teletext service.
-@item advanced_codec_digital_radio
-Advanced Codec Digital Radio service.
-@item mpeg2_digital_hdtv
-MPEG2 Digital HDTV service.
-@item advanced_codec_digital_sdtv
-Advanced Codec Digital SDTV service.
-@item advanced_codec_digital_hdtv
-Advanced Codec Digital HDTV service.
-@end table
-
 Option mpegts_flags may take a set of such flags:
 
 @table @option
@@ -756,7 +558,10 @@ Reemit PAT/PMT before writing the next packet.
 Use LATM packetization for AAC.
 @end table
 
-@subsection Example
+The recognized metadata settings in mpegts muxer are @code{service_provider}
+and @code{service_name}. If they are not set the default for
+@code{service_provider} is "FFmpeg" and the default for
+@code{service_name} is "Service01".
 
 @example
 ffmpeg -i file.mpg -c copy \
@@ -792,30 +597,6 @@ Alternatively you can write the command as:
 ffmpeg -benchmark -i INPUT -f null -
 @end example
 
-@section nut
-
-@table @option
-@item -syncpoints @var{flags}
-Change the syncpoint usage in nut:
-@table @option
-@item @var{default} use the normal low-overhead seeking aids.
-@item @var{none} do not use the syncpoints at all, reducing the overhead but making the stream non-seekable;
-    Use of this option is not recommended, as the resulting files are very damage
-    sensitive and seeking is not possible. Also in general the overhead from
-    syncpoints is negligible. Note, -@code{write_index} 0 can be used to disable
-    all growing data tables, allowing to mux endless streams with limited memory
-    and without these disadvantages.
-@item @var{timestamped} extend the syncpoint with a wallclock field.
-@end table
-The @var{none} and @var{timestamped} flags are experimental.
-@item -write_index @var{bool}
-Write index at the end, the default is to write an index.
-@end table
-
-@example
-ffmpeg -i INPUT -f_strict experimental -syncpoints none - | processor
-@end example
-
 @section ogg
 
 Ogg container muxer.
@@ -829,22 +610,15 @@ is 1 second. A value of 0 will fill all segments, making pages as large as
 possible. A value of 1 will effectively use 1 packet-per-page in most
 situations, giving a small seek granularity at the cost of additional container
 overhead.
-@item -serial_offset @var{value}
-Serial value from which to set the streams serial number.
-Setting it to different and sufficiently large values ensures that the produced
-ogg files can be safely chained.
-
 @end table
 
-@anchor{segment}
 @section segment, stream_segment, ssegment
 
 Basic stream segmenter.
 
-This muxer outputs streams to a number of separate files of nearly
-fixed duration. Output filename pattern can be set in a fashion
-similar to @ref{image2}, or by using a @code{strftime} template if
-the @option{strftime} option is enabled.
+The segmenter muxer outputs streams to a number of separate files of nearly
+fixed duration. Output filename pattern can be set in a fashion similar to
+@ref{image2}.
 
 @code{stream_segment} is a variant of the muxer used to write to
 streaming output formats, i.e. which do not require global headers,
@@ -864,21 +638,14 @@ The segment muxer works best with a single constant frame rate video.
 
 Optionally it can generate a list of the created segments, by setting
 the option @var{segment_list}. The list type is specified by the
-@var{segment_list_type} option. The entry filenames in the segment
-list are set by default to the basename of the corresponding segment
-files.
-
-See also the @ref{hls} muxer, which provides a more specific
-implementation for HLS segmentation.
-
-@subsection Options
+@var{segment_list_type} option.
 
 The segment muxer supports the following options:
 
 @table @option
 @item reference_stream @var{specifier}
 Set the reference stream, as specified by the string @var{specifier}.
-If @var{specifier} is set to @code{auto}, the reference is chosen
+If @var{specifier} is set to @code{auto}, the reference is choosen
 automatically. Otherwise it must be a stream specifier (see the ``Stream
 specifiers'' chapter in the ffmpeg manual) which specifies the
 reference stream. The default value is @code{auto}.
@@ -887,11 +654,6 @@ reference stream. The default value is @code{auto}.
 Override the inner container format, by default it is guessed by the filename
 extension.
 
-@item segment_format_options @var{options_list}
-Set output format options using a :-separated list of key=value
-parameters. Values containing the @code{:} special character must be
-escaped.
-
 @item segment_list @var{name}
 Generate also a listfile named @var{name}. If not specified no
 listfile is generated.
@@ -908,21 +670,15 @@ Allow caching (only affects M3U8 list files).
 Allow live-friendly file generation.
 @end table
 
-@item segment_list_type @var{type}
-Select the listing format.
-@table @option
-@item @var{flat} use a simple flat list of entries.
-@item @var{hls} use a m3u8-like structure.
-@end table
+Default value is @code{samp}.
 
 @item segment_list_size @var{size}
-Update the list file so that it contains at most @var{size}
+Update the list file so that it contains at most the last @var{size}
 segments. If 0 the list file will contain all the segments. Default
 value is 0.
 
-@item segment_list_entry_prefix @var{prefix}
-Prepend @var{prefix} to each entry. Useful to generate absolute paths.
-By default no prefix is applied.
+@item segment_list_type @var{type}
+Specify the format for the segment list file.
 
 The following values are recognized:
 @table @samp
@@ -973,16 +729,6 @@ Note that splitting may not be accurate, unless you force the
 reference stream key-frames at the given time. See the introductory
 notice and the examples below.
 
-@item segment_atclocktime @var{1|0}
-If set to "1" split at regular clock time intervals starting from 00:00
-o'clock. The @var{time} value specified in @option{segment_time} is
-used for setting the length of the splitting interval.
-
-For example with @option{segment_time} set to "900" this makes it possible
-to create files at 12:00 o'clock, 12:15, 12:30, etc.
-
-Default value is "0".
-
 @item segment_time_delta @var{delta}
 Specify the accuracy time when selecting the start time for a
 segment, expressed as a duration specification. Default value is "0".
@@ -1002,7 +748,7 @@ In particular may be used in combination with the @file{ffmpeg} option
 @var{force_key_frames} may not be set accurately because of rounding
 issues, with the consequence that a key frame time may result set just
 before the specified time. For constant frame rate videos a value of
-1/(2*@var{frame_rate}) should address the worst case mismatch between
+1/2*@var{frame_rate} should address the worst case mismatch between
 the specified time and the time set by @var{force_key_frames}.
 
 @item segment_times @var{times}
@@ -1024,12 +770,6 @@ Wrap around segment index once it reaches @var{limit}.
 @item segment_start_number @var{number}
 Set the sequence number of the first segment. Defaults to @code{0}.
 
-@item strftime @var{1|0}
-Use the @code{strftime} function to define the name of the new
-segments to write. If this is selected, the output segment name must
-contain a @code{strftime} function template. Default value is
-@code{0}.
-
 @item reset_timestamps @var{1|0}
 Reset timestamps at the begin of each segment, so that each segment
 will start with near-zero timestamps. It is meant to ease the playback
@@ -1045,7 +785,7 @@ argument must be a time duration specification, and defaults to 0.
 
 @itemize
 @item
-Remux the content of file @file{in.mkv} to a list of segments
+To remux the content of file @file{in.mkv} to a list of segments
 @file{out-000.nut}, @file{out-001.nut}, etc., and write the list of
 generated segments to @file{out.list}:
 @example
@@ -1053,20 +793,14 @@ ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.list out%03d.nu
 @end example
 
 @item
-Segment input and set output format options for the output segments:
-@example
-ffmpeg -i in.mkv -f segment -segment_time 10 -segment_format_options movflags=+faststart out%03d.mp4
-@end example
-
-@item
-Segment the input file according to the split points specified by the
-@var{segment_times} option:
+As the example above, but segment the input file according to the split
+points specified by the @var{segment_times} option:
 @example
 ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 out%03d.nut
 @end example
 
 @item
-Use the @command{ffmpeg} @option{force_key_frames}
+As the example above, but use the @command{ffmpeg} @option{force_key_frames}
 option to force key frames in the input at the specified location, together
 with the segment option @option{segment_time_delta} to account for
 possible roundings operated when setting key frame times.
@@ -1085,7 +819,7 @@ ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_fr
 @end example
 
 @item
-Convert the @file{in.mkv} to TS segments using the @code{libx264}
+To convert the @file{in.mkv} to TS segments using the @code{libx264}
 and @code{libfaac} encoders:
 @example
 ffmpeg -i in.mkv -map 0 -codec:v libx264 -codec:a libfaac -f ssegment -segment_list out.list out%03d.ts
@@ -1100,28 +834,6 @@ ffmpeg -re -i in.mkv -codec copy -map 0 -f segment -segment_list playlist.m3u8 \
 @end example
 @end itemize
 
-@section smoothstreaming
-
-Smooth Streaming muxer generates a set of files (Manifest, chunks) suitable for serving with conventional web server.
-
-@table @option
-@item window_size
-Specify the number of fragments kept in the manifest. Default 0 (keep all).
-
-@item extra_window_size
-Specify the number of fragments kept outside of the manifest before removing from disk. Default 5.
-
-@item lookahead_count
-Specify the number of lookahead fragments. Default 2.
-
-@item min_frag_duration
-Specify the minimum fragment duration (in microseconds). Default 5000000.
-
-@item remove_at_exit
-Specify whether to remove all fragments when finished. Default 0 (do not remove).
-
-@end table
-
 @section tee
 
 The tee muxer can be used to write the same data to several files or any
@@ -1137,8 +849,8 @@ to feed the same packets to several muxers directly.
 The slave outputs are specified in the file name given to the muxer,
 separated by '|'. If any of the slave name contains the '|' separator,
 leading or trailing spaces or any special character, it must be
-escaped (see @ref{quoting_and_escaping,,the "Quoting and escaping"
-section in the ffmpeg-utils(1) manual,ffmpeg-utils}).
+escaped (see the ``Quoting and escaping'' section in the ffmpeg-utils
+manual).
 
 Muxer options can be specified for each slave by prepending them as a list of
 @var{key}=@var{value} pairs separated by ':', between square brackets. If
@@ -1153,13 +865,10 @@ output name suffix.
 
 @item bsfs[/@var{spec}]
 Specify a list of bitstream filters to apply to the specified
-output.
-
-It is possible to specify to which streams a given bitstream filter
-applies, by appending a stream specifier to the option separated by
-@code{/}. @var{spec} must be a stream specifier (see @ref{Format
-stream specifiers}).  If the stream specifier is not specified, the
-bitstream filters will be applied to all streams in the output.
+output. It is possible to specify to which streams a given bitstream
+filter applies, by appending a stream specifier to the option
+separated by @code{/}. If the stream specifier is not specified, the
+bistream filters will be applied to all streams in the output.
 
 Several bitstream filters can be specified, separated by ",".
 
@@ -1169,8 +878,7 @@ specified by a stream specifier. If not specified, this defaults to
 all the input streams.
 @end table
 
-@subsection Examples
-
+Some examples follow.
 @itemize
 @item
 Encode something and both archive it in a WebM file and stream it
@@ -1191,49 +899,10 @@ audio packets.
 ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac -strict experimental
        -f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=a]out.aac"
 @end example
-
-@item
-As below, but select only stream @code{a:1} for the audio output. Note
-that a second level escaping must be performed, as ":" is a special
-character used to separate options.
-@example
-ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac -strict experimental
-       -f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=\'a:1\']out.aac"
-@end example
 @end itemize
 
 Note: some codecs may need different options depending on the output format;
 the auto-detection of this can not work with the tee muxer. The main example
 is the @option{global_header} flag.
 
-@section webm_dash_manifest
-
-WebM DASH Manifest muxer.
-
-This muxer implements the WebM DASH Manifest specification to generate the DASH manifest XML.
-
-@subsection Options
-
-This muxer supports the following options:
-
-@table @option
-@item adaptation_sets
-This option has the following syntax: "id=x,streams=a,b,c id=y,streams=d,e" where x and y are the
-unique identifiers of the adaptation sets and a,b,c,d and e are the indices of the corresponding
-audio and video streams. Any number of adaptation sets can be added using this option.
-@end table
-
-@subsection Example
-@example
-ffmpeg -f webm_dash_manifest -i video1.webm \
-       -f webm_dash_manifest -i video2.webm \
-       -f webm_dash_manifest -i audio1.webm \
-       -f webm_dash_manifest -i audio2.webm \
-       -map 0 -map 1 -map 2 -map 3 \
-       -c copy \
-       -f webm_dash_manifest \
-       -adaptation_sets "id=0,streams=0,1 id=1,streams=2,3" \
-       manifest.xml
-@end example
-
 @c man end MUXERS

doc/nut.texi

@@ -1,5 +1,4 @@
 \input texinfo @c -*- texinfo -*-
-@documentencoding UTF-8
 
 @settitle NUT
 
@@ -22,27 +21,6 @@ The official nut specification is at svn://svn.mplayerhq.hu/nut
 In case of any differences between this text and the official specification,
 the official specification shall prevail.
 
-@chapter Modes
-NUT has some variants signaled by using the flags field in its main header.
-
-@multitable @columnfractions .4 .4
-@item BROADCAST   @tab Extend the syncpoint to report the sender wallclock
-@item PIPE        @tab Omit completely the syncpoint
-@end multitable
-
-@section BROADCAST
-
-The BROADCAST variant provides a secondary time reference to facilitate
-detecting endpoint latency and network delays.
-It assumes all the endpoint clocks are syncronized.
-To be used in real-time scenarios.
-
-@section PIPE
-
-The PIPE variant assumes NUT is used as non-seekable intermediate container,
-by not using syncpoint removes unneeded overhead and reduces the overall
-memory usage.
-
 @chapter Container-specific codec tags
 
 @section Generic raw YUVA formats